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		<updated>2026-06-28T04:59:49Z</updated>
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				<updated>2021-07-16T15:56:26Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/Servers</id>
		<title>Servers</title>
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				<updated>2021-07-16T15:54:07Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;[[File:ChooseServerImg2.png|thumb|none|1280px|VoIP.ms servers]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
{| class=&amp;quot;wikitable&amp;quot;&lt;br /&gt;
|+&lt;br /&gt;
|-&lt;br /&gt;
! Article en Français !! Artículo en Español&lt;br /&gt;
|-&lt;br /&gt;
| [[https://wiki.voip.ms/article/Choisir_un_serveur Français]] || &lt;br /&gt;
[[https://wiki.voip.ms/article/Elegir_servidor Español]] &lt;br /&gt;
|}&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
= Choosing a Server =&lt;br /&gt;
&lt;br /&gt;
[http://www.voip.ms VoIP.ms] offers many different servers, but which one should you choose? One misconception is that you should pick the closest to your location, however this is not needed most of the time. For example, if you are in the USA, any of the US servers will provide a really good latency and service quality. The newest server within a city is indicated with the highest number attached to the name, as they are classified in ascending order. Also worth noting is that there is a network tool that will help you when deciding which server you want to use, generally named a &amp;quot;ping&amp;quot;, which will provide you the latency between you and the server. Therefore the server which provides you less latency should be used.&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' Please bear in mind that some servers might not be available for your DID number to be used as POP (Point of presence) at the ''Manage DIDs'' section. &lt;br /&gt;
 Make sure that your SIP/IAX device and your phone number are pointing to the same server. &lt;br /&gt;
&lt;br /&gt;
=== IPs ===&lt;br /&gt;
&lt;br /&gt;
&amp;lt;div&amp;gt;&amp;lt;ul&amp;gt; &lt;br /&gt;
&amp;lt;li style=&amp;quot;display: inline-block; vertical-align:top;&amp;quot;&amp;gt; &lt;br /&gt;
'''Canada'''&lt;br /&gt;
*Montreal 1, QC     ('''montreal.voip.ms''')    192.175.96.73&lt;br /&gt;
*Montreal 2, QC     ('''montreal2.voip.ms''')   192.175.96.74&lt;br /&gt;
*Montreal 3, QC     ('''montreal3.voip.ms''')   192.175.96.68&lt;br /&gt;
*Montreal 4, QC     ('''montreal4.voip.ms''')   67.205.74.179&lt;br /&gt;
*Montreal 5, QC     ('''montreal5.voip.ms''')   192.175.96.69&lt;br /&gt;
*Montreal 6, QC     ('''montreal6.voip.ms''')   192.175.96.70&lt;br /&gt;
*Montreal 7, QC     ('''montreal7.voip.ms''')   192.175.96.71&lt;br /&gt;
*Montreal 8, QC     ('''montreal8.voip.ms''')   192.175.96.72&lt;br /&gt;
*Montreal 9, QC     ('''montreal9.voip.ms''')   67.205.74.184&lt;br /&gt;
*Montreal 10, QC     ('''montreal10.voip.ms''') 67.205.74.187&lt;br /&gt;
*Toronto 1, ON      ('''toronto.voip.ms''')     158.85.70.148&lt;br /&gt;
*Toronto 2, ON      ('''toronto2.voip.ms''')    158.85.70.149&lt;br /&gt;
*Toronto 3, ON      ('''toronto3.voip.ms''')    158.85.70.150&lt;br /&gt;
*Toronto 4, ON      ('''toronto4.voip.ms''')    158.85.70.151&lt;br /&gt;
*Toronto 5, ON      ('''toronto5.voip.ms''')    184.75.215.106&lt;br /&gt;
*Toronto 6, ON      ('''toronto6.voip.ms''')    184.75.215.114&lt;br /&gt;
*Toronto 7, ON      ('''toronto7.voip.ms''')    184.75.215.146&lt;br /&gt;
*Toronto 8, ON      ('''toronto8.voip.ms''')    184.75.213.210&lt;br /&gt;
*Toronto 9, ON      ('''toronto9.voip.ms''')    158.85.70.154&lt;br /&gt;
*Toronto 10, ON      ('''toronto10.voip.ms''')    158.85.70.158&lt;br /&gt;
*Vancouver 1, BC    ('''vancouver.voip.ms''')   162.213.157.220&lt;br /&gt;
*Vancouver 2, BC    ('''vancouver2.voip.ms''')  162.213.157.117&lt;br /&gt;
*Vancouver 3, BC    ('''vancouver3.voip.ms''')  162.213.157.82&lt;br /&gt;
&amp;lt;/li&amp;gt;&lt;br /&gt;
&lt;br /&gt;
&amp;lt;li style=&amp;quot;display: inline-block; vertical-align:top;&amp;quot;&amp;gt; &lt;br /&gt;
'''United States'''&lt;br /&gt;
*Atlanta 1, GA      ('''atlanta.voip.ms''')     75.127.65.130&lt;br /&gt;
*Atlanta 2, GA      ('''atlanta2.voip.ms''')    209.217.224.50&lt;br /&gt;
*Chicago 1, IL      ('''chicago.voip.ms''')     69.162.175.27&lt;br /&gt;
*Chicago 2, IL      ('''chicago2.voip.ms''')    69.162.175.28 &lt;br /&gt;
*Chicago 3, IL      ('''chicago3.voip.ms''')    69.162.175.29&lt;br /&gt;
*Chicago 4, IL      ('''chicago4.voip.ms''')    208.100.39.55&lt;br /&gt;
*Chicago 5, IL      ('''chicago5.voip.ms''')    50.31.115.149&lt;br /&gt;
*Chicago 6, IL      ('''chicago6.voip.ms''')    50.31.115.150&lt;br /&gt;
*Chicago 7, IL      ('''chicago7.voip.ms''')    50.31.115.151&lt;br /&gt;
*Dallas, TX         ('''dallas.voip.ms''')      158.85.149.162&lt;br /&gt;
*Dallas 2, TX         ('''dallas2.voip.ms''')   158.85.149.163&lt;br /&gt;
*Denver 1, CO       ('''denver.voip.ms''')      23.239.211.90 &lt;br /&gt;
*Denver 2, CO       ('''denver2.voip.ms''')     64.27.52.226&lt;br /&gt;
*Houston, TX        ('''houston.voip.ms''')     173.193.85.18&lt;br /&gt;
*Houston 2, TX        ('''houston2.voip.ms''')  173.193.85.19&lt;br /&gt;
*Los Angeles 1, CA  ('''losangeles.voip.ms''')  96.44.149.186&lt;br /&gt;
*Los Angeles 2, CA  ('''losangeles2.voip.ms''') 96.44.149.202&lt;br /&gt;
*Los Angeles 3, CA  ('''losangeles3.voip.ms''') 64.188.6.162&lt;br /&gt;
*Los Angeles 4, CA  ('''losangeles4.voip.ms''') 64.188.6.170&lt;br /&gt;
*New York 1, NY     ('''newyork.voip.ms''')     72.251.239.196&lt;br /&gt;
*New York 2, NY     ('''newyork2.voip.ms''')    72.251.239.205&lt;br /&gt;
*New York 3, NY     ('''newyork3.voip.ms''')    72.251.239.206&lt;br /&gt;
*New York 4, NY     ('''newyork4.voip.ms''')    72.251.239.207&lt;br /&gt;
*New York 5, NY     ('''newyork5.voip.ms''')    23.29.136.28&lt;br /&gt;
*New York 6, NY     ('''newyork6.voip.ms''')    23.29.136.29&lt;br /&gt;
*New York 7, NY     ('''newyork7.voip.ms''')    23.29.136.38&lt;br /&gt;
*New York 8, NY     ('''newyork8.voip.ms''')    23.29.136.40 &lt;br /&gt;
*San Jose, CA       ('''sanjose.voip.ms''')     23.246.247.146&lt;br /&gt;
*San Jose 2, CA     ('''sanjose2.voip.ms''')    23.246.247.147&lt;br /&gt;
*Seattle 1, WA      ('''seattle.voip.ms''')     50.23.160.53&lt;br /&gt;
*Seattle 2, WA      ('''seattle2.voip.ms''')    50.23.149.166&lt;br /&gt;
*Seattle 3, WA      ('''seattle3.voip.ms''')    50.23.160.54&lt;br /&gt;
*Tampa, FL          ('''tampa.voip.ms''')       162.254.144.173&lt;br /&gt;
*Tampa 2, FL        ('''tampa2.voip.ms''')      162.254.144.176&lt;br /&gt;
*Tampa 3, FL        ('''tampa3.voip.ms''')      23.111.187.139&lt;br /&gt;
*Tampa 4, FL        ('''tampa4.voip.ms''')      23.111.166.202&lt;br /&gt;
*Washington 1, DC   ('''washington.voip.ms''')  208.43.234.226&lt;br /&gt;
*Washington 2, DC   ('''washington2.voip.ms''') 208.43.234.227&lt;br /&gt;
&amp;lt;/li&amp;gt;&lt;br /&gt;
&lt;br /&gt;
&amp;lt;li style=&amp;quot;display: inline-block; vertical-align:top;&amp;quot;&amp;gt; &lt;br /&gt;
'''International'''&lt;br /&gt;
*Amsterdam, NL      ('''amsterdam.voip.ms''')   66.212.22.42&lt;br /&gt;
*London, UK         ('''london.voip.ms''')      159.8.157.212&lt;br /&gt;
*Paris, FR          ('''paris.voip.ms''')       159.8.85.180&lt;br /&gt;
&amp;lt;/li&amp;gt;&lt;br /&gt;
&amp;lt;/ul&amp;gt;&amp;lt;/div&amp;gt;&lt;br /&gt;
&lt;br /&gt;
===Server Realms===&lt;br /&gt;
&lt;br /&gt;
For IOS, Please click here [http://wiki.voip.ms/article/Server_Realms Server Realms] to get the Realm Name for the server you plan on using, this can differ from the Domain Name being used. &lt;br /&gt;
&lt;br /&gt;
= What is a Ping? =&lt;br /&gt;
&lt;br /&gt;
Ping is a standard tool used to test network connections. It is mostly used to determine if a server or device can be reached across the network and the latency of the response(the time it takes to send a packet to the destination and for it to return to your computer).&lt;br /&gt;
&lt;br /&gt;
Ping tools are part of Windows, Mac OS X and Linux as well as some routers.&lt;br /&gt;
&lt;br /&gt;
== How does the ping work? ==&lt;br /&gt;
&lt;br /&gt;
It sends request messages to a target network address or DNS names at periodic intervals and measures the time it takes for a response message to arrive and return(better known as latency). &lt;br /&gt;
&lt;br /&gt;
==How to ping on a PC==&lt;br /&gt;
&lt;br /&gt;
Pinging is a command which tells you if the connection between your computer and a particular domain is working correctly.&lt;br /&gt;
&lt;br /&gt;
In Windows, select Start &amp;gt; Programs &amp;gt; Accessories &amp;gt; Command Prompt. This will give you a window like the one below.&lt;br /&gt;
&lt;br /&gt;
Enter the word ping, followed by a space, then the domain name.(montreal.voip.ms) in this case domain is our server name.&lt;br /&gt;
&lt;br /&gt;
If the results show a series of replies, the connection is working. The time shows you how fast the connection is. If you see a &amp;quot;timed out&amp;quot; error instead of a reply, there is a breakdown somewhere between your computer and the domain.&lt;br /&gt;
&lt;br /&gt;
[[File:Ping.gif|thumb|none|600px|Ping]]&lt;br /&gt;
&lt;br /&gt;
==How to ping on a Mac Computer==&lt;br /&gt;
&lt;br /&gt;
1- Click on Finder in the dock.&lt;br /&gt;
&lt;br /&gt;
2- Click on Applications.&lt;br /&gt;
&lt;br /&gt;
3- Click on Utilities.&lt;br /&gt;
&lt;br /&gt;
4- Double-click on Network Utility. &amp;amp;#42;&lt;br /&gt;
&lt;br /&gt;
&amp;amp;#42; In OS X Mavericks (10.9.x) this utility app changed location. Launch it from spotlight instead, either press &amp;quot;command&amp;quot;+&amp;quot;space bar&amp;quot; or click on spotlight directly (magnifying glass icon at top right of screen), type &amp;quot;network utility&amp;quot; and hit &amp;quot;return&amp;quot;&lt;br /&gt;
&lt;br /&gt;
5- In the Network Utility window, click on the Ping tab&lt;br /&gt;
&lt;br /&gt;
6- In the field under &amp;quot;Please enter the network address to ping,&amp;quot; like montreal.voip.ms&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
''If pings results are not consistent, you may have an issue with Jitter. You can work on this issue by adjusting the &amp;quot;Network Jitter Level&amp;quot; setting on your VoIP device. Usually a ping of under 150 ms is recommended in order to have good quality. The latency time to the server is important, however there are also other factors that could affect the quality of the calls such as packet loss (VoIP communications are very sensitive to this), and the Jitter level of your Internet connection.''&lt;br /&gt;
&lt;br /&gt;
The following is the output of running ping with the target losangeles.voip.ms.&lt;br /&gt;
&lt;br /&gt;
 #ping losangeles.voip.ms&lt;br /&gt;
 Ping to losangeles.voip.ms [67.215.241.250] with 32 bytes de datos:&lt;br /&gt;
 Response from 67.215.241.250: bytes=32 time=67ms TTL=52&lt;br /&gt;
 Response from 67.215.241.250: bytes=32 time=69ms TTL=52&lt;br /&gt;
 Response from 67.215.241.250: bytes=32 time=68ms TTL=52&lt;br /&gt;
 Response from 67.215.241.250: bytes=32 time=67ms TTL=52&lt;br /&gt;
 ping statistics from 67.215.241.250:&lt;br /&gt;
 4 packets transmitted, 4 received, 0% packet lost. rtt min/avg/max/mdev = 67ms, 69ms, 67ms&lt;br /&gt;
&lt;br /&gt;
Sample ping output in windows:&lt;br /&gt;
 C:\Windows\system32&amp;gt;ping montreal.voip.ms&lt;br /&gt;
 &lt;br /&gt;
 Pinging montreal.voip.ms [67.205.74.184] with 32 bytes of data:&lt;br /&gt;
 Reply from 67.205.74.184: bytes=32 time=85ms TTL=49&lt;br /&gt;
 Reply from 67.205.74.184: bytes=32 time=86ms TTL=49&lt;br /&gt;
 Reply from 67.205.74.184: bytes=32 time=86ms TTL=49&lt;br /&gt;
 Reply from 67.205.74.184: bytes=32 time=85ms TTL=49&lt;br /&gt;
 &lt;br /&gt;
 Ping statistics for 67.205.74.184:&lt;br /&gt;
     Packets: Sent = 4, Received = 4, Lost = 0 (0% loss),&lt;br /&gt;
 Approximate round trip times in milli-seconds:&lt;br /&gt;
     Minimum = 85ms, Maximum = 86ms, Average = 85ms&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
= Latency Testing Scripts (User Submitted) =&lt;br /&gt;
&amp;lt;p&amp;gt;All the following scripts were produced by voip.ms users who felt others might also benefit from the output of their efforts.  They were written over a span of Years and probably need adjusting before you use them, to cater for changes in servers over time and changes in policies (like not testing heavily subscribed servers which are not open to new registrations)&amp;lt;/p&amp;gt;&lt;br /&gt;
&amp;lt;p&amp;gt;If you aren't satisfied that the scripts are safe or simply don’t like the way they syntactically appear, you can still manually ping a selection of servers and choose a server based on the best latency. The following scripts are essentially just wrappers around the ping command which support lists of servers to feed to ping and present the output in a readable format.&amp;lt;/p&amp;gt;&lt;br /&gt;
&amp;lt;p&amp;gt;If you feel you have a simpler cleaner script that works for another platform or language, please do add it to this wiki via a support ticket.&lt;br /&gt;
&amp;lt;/p&amp;gt;&lt;br /&gt;
=== Bash Script To Handle The Mac Ping Output Format ===&lt;br /&gt;
&lt;br /&gt;
&amp;lt;p&amp;gt;To make use of this script (1) save as a plain text file (2) set permissions on the file to executable (3) invoke script&lt;br /&gt;
e.g. Save script below using your favourite editor as pingVoipMS.sh (2) chmod u+x pingVoipMS.sh (3) ./pingVoipMS.sh&lt;br /&gt;
This is a bash 3.x script, so it also works in Linux, just change the ping packet loss field from 7 to 6 in the final loop below (or wherever the loss field is in your ping output format).  Depending upon your distro curl might need to change to wget.&lt;br /&gt;
&amp;lt;/p&amp;gt;&lt;br /&gt;
&amp;lt;pre&amp;gt;&lt;br /&gt;
#!/bin/bash&lt;br /&gt;
# Ping several servers and display Latency, Jitter and Packet Loss&lt;br /&gt;
#      Usage: [-c &amp;lt;count&amp;gt;][-i &amp;lt;wait time&amp;gt;][-r test restricted servers][&amp;lt;server list file&amp;gt;]&lt;br /&gt;
#&lt;br /&gt;
# The optional server list text file should be formatted with one host name per line.&lt;br /&gt;
# The list of voip.ms servers is available at https://wiki.voip.ms/article/Choosing_Server&lt;br /&gt;
# If no args are supplied, this script will scrape a ping server list from voip.ms&lt;br /&gt;
#&lt;br /&gt;
USER_FILE=&amp;quot;&amp;quot;&lt;br /&gt;
COUNT=3; INTERVAL=5; RESTRICTED=0&lt;br /&gt;
restrictedList=(atlanta.voip.ms chicago.voip.ms&lt;br /&gt;
                montreal.voip.ms montreal2.voip.ms montreal3.voip.ms montreal4.voip.ms&lt;br /&gt;
                newyork.voip.ms newyork4.voip.ms seattle.voip.ms&lt;br /&gt;
                toronto.voip.ms toronto2.voip.ms toronto3.voip.ms toronto4.voip.ms)&lt;br /&gt;
&lt;br /&gt;
# Handle any passed in script arguments&lt;br /&gt;
while getopts c:i:r parm&lt;br /&gt;
do&lt;br /&gt;
    case $parm in&lt;br /&gt;
        c)count_opt=$OPTARG;;&lt;br /&gt;
        i)interval_opt=$OPTARG;;&lt;br /&gt;
        r)RESTRICTED=1;;&lt;br /&gt;
        *)echo -e &amp;quot;Invalid arg\nUsage:\t[ -c &amp;lt;count of ECHO_REQUESTs to Tx, default 3&amp;gt; ] \&lt;br /&gt;
                  \n\t[ -i &amp;lt;wait time (s) between datagrams, default 5&amp;gt; ]                \&lt;br /&gt;
                  \n\t[ -r ] Include restricted servers in latency test                  \&lt;br /&gt;
                  \n\t[FILE &amp;lt;ping server list&amp;gt; ]&amp;quot;;exit 1;;&lt;br /&gt;
    esac&lt;br /&gt;
done&lt;br /&gt;
&lt;br /&gt;
# Test if an option was specified and whether it's a +ve non-zero integer&lt;br /&gt;
[[ -n $count_opt    &amp;amp;&amp;amp; ($count_opt =~ ^[[:digit:]]+$)    &amp;amp;&amp;amp; $count_opt -gt 0 ]] &amp;amp;&amp;amp;&lt;br /&gt;
        COUNT=$count_opt&lt;br /&gt;
[[ -n $interval_opt &amp;amp;&amp;amp; ($interval_opt =~ ^[[:digit:]]+$) &amp;amp;&amp;amp; $interval_opt -gt 0 ]] &amp;amp;&amp;amp;&lt;br /&gt;
        INTERVAL=$interval_opt&lt;br /&gt;
&lt;br /&gt;
shift $((OPTIND - 1))&lt;br /&gt;
&lt;br /&gt;
# Validate supplied file (server list)&lt;br /&gt;
[[ -n $1 &amp;amp;&amp;amp; ! (-f $1 &amp;amp;&amp;amp; -r $1) ]] &amp;amp;&amp;amp;&lt;br /&gt;
        { echo &amp;quot;\&amp;quot;$1\&amp;quot; file does not exist or is not readable&amp;quot;; exit 1; }&lt;br /&gt;
[[ -n $1 &amp;amp;&amp;amp; -f $1 &amp;amp;&amp;amp; -r $1 ]] &amp;amp;&amp;amp; USER_FILE=&amp;quot;$1&amp;quot;&lt;br /&gt;
&lt;br /&gt;
if [[ -n $USER_FILE ]]&lt;br /&gt;
then&lt;br /&gt;
# Bash 3.x in macOS does not support readarray, need to do cumbersome array loops instead&lt;br /&gt;
    while IFS= read -r servers; do&lt;br /&gt;
        serverList+=( &amp;quot;$servers&amp;quot; )&lt;br /&gt;
    done &amp;lt; &amp;lt;(grep -Eo '^\b[[:alpha:]]+?[[:alnum:]]\.voip\.ms\b$' &amp;quot;$USER_FILE&amp;quot; | \&lt;br /&gt;
             grep -v '^\s*#' | awk NF | sort)&lt;br /&gt;
else&lt;br /&gt;
# N.B. The script looks for the html boldface tags &amp;lt;b&amp;gt; &amp;lt;/b&amp;gt; inside a bracket&lt;br /&gt;
# If the website alters and the parse fails, manually create the list and&lt;br /&gt;
# supply as a script arg (or perhaps update the parsing to work again :)&lt;br /&gt;
    while IFS= read -r servers; do&lt;br /&gt;
        serverList+=( &amp;quot;$servers&amp;quot; )&lt;br /&gt;
    done &amp;lt; &amp;lt;(curl -sm 10 https://wiki.voip.ms/article/Choosing_Server | \&lt;br /&gt;
             grep -E '(&amp;lt;b&amp;gt;[[:alpha:]]+?[[:alnum:]]\.voip\.ms&amp;lt;/b&amp;gt;)'    | \&lt;br /&gt;
             tr &amp;quot;&amp;lt;&amp;gt;&amp;quot; &amp;quot; &amp;quot; | awk '{print $(NF-3)}' | sort                 )&lt;br /&gt;
fi&lt;br /&gt;
&lt;br /&gt;
# Newer voip.ms clients can't register onto these over-subscribed servers&lt;br /&gt;
# Don't test the restricted list unless explicitly asked (with the -r cmd line option)&lt;br /&gt;
if [[ $RESTRICTED -eq 0 ]]&lt;br /&gt;
then&lt;br /&gt;
    for server in &amp;quot;${restrictedList[@]}&amp;quot;&lt;br /&gt;
    do&lt;br /&gt;
        ix=$(printf &amp;quot;%s\n&amp;quot; &amp;quot;${serverList[@]}&amp;quot; | grep -n &amp;quot;^${server}&amp;quot; | cut -d &amp;quot;:&amp;quot; -f1)&lt;br /&gt;
        while IFS= read -ra idx; do&lt;br /&gt;
            keys+=( &amp;quot;${idx[@]}&amp;quot; )&lt;br /&gt;
        done &amp;lt; &amp;lt;([[ $ix -gt 0 ]] &amp;amp;&amp;amp; echo $((ix-1)))&lt;br /&gt;
    done&lt;br /&gt;
    for ((i=${#keys[@]} - 1; i &amp;gt;= 0; i--)); do unset &amp;quot;serverList[keys[i]]&amp;quot;; done&lt;br /&gt;
fi&lt;br /&gt;
&lt;br /&gt;
if [[ ${#serverList[@]} -eq 0 ]]&lt;br /&gt;
then&lt;br /&gt;
    echo &amp;quot;No unrestricted Voip.ms servers could be found, please supply a server list&amp;quot;&lt;br /&gt;
    exit 1&lt;br /&gt;
fi&lt;br /&gt;
&lt;br /&gt;
runTime=$((COUNT * INTERVAL * ${#serverList[@]}))&lt;br /&gt;
&lt;br /&gt;
echo &amp;quot;PING will send $COUNT packet(s) with a wait of $INTERVAL sec(s) between each packet&amp;quot;&lt;br /&gt;
echo &amp;quot;Change the PING options by invoking this script with -c and/or -i, default \&amp;quot;-c 3 -i 5\&amp;quot;&amp;quot;&lt;br /&gt;
echo &amp;quot;Over $((${#serverList[@]})) server(s) the estimated script Run Time will be $runTime seconds&amp;quot;&lt;br /&gt;
echo &amp;quot;================================================================&amp;quot;&lt;br /&gt;
printf &amp;quot;%-20s %-18s %7s %8s %6s   %s\n&amp;quot; &amp;quot;VoIP Server&amp;quot; &amp;quot;IP Address&amp;quot; &amp;quot;Latency&amp;quot; &amp;quot;Jitter&amp;quot; &amp;quot;Loss&amp;quot; &amp;quot;Countdown&amp;quot;&lt;br /&gt;
echo &amp;quot;================================================================  (seconds)&amp;quot;&lt;br /&gt;
&lt;br /&gt;
for myLn in &amp;quot;${serverList[@]}&amp;quot;&lt;br /&gt;
do&lt;br /&gt;
     while IFS=$'\n' read -r pings; do&lt;br /&gt;
         pingList+=( &amp;quot;$pings&amp;quot; )&lt;br /&gt;
         printf &amp;quot;%-64s %5d   %2d/%-2d\n&amp;quot; &amp;quot;$pings&amp;quot; \&lt;br /&gt;
                &amp;quot;$((runTime - COUNT * INTERVAL * ${#pingList[@]}))&amp;quot; \&lt;br /&gt;
                &amp;quot;${#pingList[@]}&amp;quot; &amp;quot;${#serverList[@]}&amp;quot;&lt;br /&gt;
     done &amp;lt; &amp;lt;( ping -c &amp;quot;$COUNT&amp;quot; -i &amp;quot;$INTERVAL&amp;quot; -q &amp;quot;$myLn&amp;quot; | awk \&lt;br /&gt;
     '&lt;br /&gt;
        /^PING / {myH=$2}&lt;br /&gt;
        /^PING / {&lt;br /&gt;
            IP = substr($3,2,15)&lt;br /&gt;
            split(IP,myIP,&amp;quot;)&amp;quot;) }&lt;br /&gt;
        /packet loss/ {myPL=$7}&lt;br /&gt;
        /min\/avg\/max/ {&lt;br /&gt;
            split($4,myS,&amp;quot;/&amp;quot;)&lt;br /&gt;
            printf(&amp;quot;%-20s %-18s %7.3f %8.3f %6s\n&amp;quot;,&lt;br /&gt;
                    myH, myIP[1], myS[2], myS[4], myPL ) }&lt;br /&gt;
     ' )&lt;br /&gt;
done&lt;br /&gt;
&lt;br /&gt;
echo &amp;quot;================================================================&amp;quot;&lt;br /&gt;
echo -e &amp;quot;\nMost appropriate server listed in order of best latency\n&amp;quot;&lt;br /&gt;
echo &amp;quot;================================================================&amp;quot;&lt;br /&gt;
printf &amp;quot;%-20s %-18s %7s %8s %6s\n&amp;quot; &amp;quot;VoIP Server&amp;quot; &amp;quot;IP Address&amp;quot; &amp;quot;Latency&amp;quot; &amp;quot;Jitter&amp;quot; &amp;quot;Loss&amp;quot;&lt;br /&gt;
echo &amp;quot;================================================================&amp;quot;&lt;br /&gt;
printf &amp;quot;%s\n&amp;quot; &amp;quot;${pingList[@]}&amp;quot; | LC_ALL=C sort -n -k 3,3 -k 5,5 -k 4,4 | \&lt;br /&gt;
        awk '{printf(&amp;quot;%s    \(%2d\)\n&amp;quot;,$0, NR)}'&lt;br /&gt;
echo &amp;quot;================================================================&amp;quot;&lt;br /&gt;
&amp;lt;/pre&amp;gt;&lt;br /&gt;
&lt;br /&gt;
=== Perl Script ===&lt;br /&gt;
Pings list of voip.ms servers round robin with optional output csv file.&lt;br /&gt;
&lt;br /&gt;
    # usage ping_voip.ms.pl &amp;lt;number of times&amp;gt; &amp;lt;seconds in between&amp;gt; &amp;lt;output.csv&amp;gt;&lt;br /&gt;
    use Net::Ping;&lt;br /&gt;
    use Time::HiRes;&lt;br /&gt;
    use strict;&lt;br /&gt;
    &lt;br /&gt;
    # input list &lt;br /&gt;
    my @hosts = qw(&lt;br /&gt;
        atlanta.voip.ms&lt;br /&gt;
        atlanta2.voip.ms&lt;br /&gt;
        chicago.voip.ms&lt;br /&gt;
        chicago2.voip.ms&lt;br /&gt;
        chicago3.voip.ms&lt;br /&gt;
        chicago4.voip.ms&lt;br /&gt;
        dallas.voip.ms&lt;br /&gt;
        denver.voip.ms&lt;br /&gt;
        denver2.voip.ms&lt;br /&gt;
        houston.voip.ms&lt;br /&gt;
        losangeles.voip.ms&lt;br /&gt;
        losangeles2.voip.ms&lt;br /&gt;
        newyork.voip.ms&lt;br /&gt;
        newyork2.voip.ms&lt;br /&gt;
        newyork3.voip.ms&lt;br /&gt;
        newyork4.voip.ms&lt;br /&gt;
        seattle.voip.ms&lt;br /&gt;
        seattle2.voip.ms&lt;br /&gt;
        seattle3.voip.ms&lt;br /&gt;
        tampa.voip.ms&lt;br /&gt;
        washington.voip.ms&lt;br /&gt;
        washington2.voip.ms&lt;br /&gt;
        montreal.voip.ms&lt;br /&gt;
        montreal2.voip.ms&lt;br /&gt;
        montreal3.voip.ms&lt;br /&gt;
        montreal4.voip.ms&lt;br /&gt;
        toronto2.voip.ms&lt;br /&gt;
        toronto3.voip.ms&lt;br /&gt;
        toronto4.voip.ms&lt;br /&gt;
        toronto.voip.ms&lt;br /&gt;
        london.voip.ms&lt;br /&gt;
    );&lt;br /&gt;
    &lt;br /&gt;
    $| = 1; #autoflush&lt;br /&gt;
    # High precision syntax (requires Time::HiRes)&lt;br /&gt;
    my $p = Net::Ping-&amp;gt;new(&amp;quot;icmp&amp;quot;,1);&lt;br /&gt;
    $p-&amp;gt;hires();&lt;br /&gt;
    my $max_name_length = (reverse sort { $a &amp;lt;=&amp;gt; $b } map { length($_) } @hosts)[0];&lt;br /&gt;
    my $count = 4; # number of times to ping&lt;br /&gt;
    my $interval = 5; # seconds between ping rounds&lt;br /&gt;
    my $output_file = &amp;quot;&amp;quot;;&lt;br /&gt;
    my @data;&lt;br /&gt;
    &lt;br /&gt;
    # check for arguments&lt;br /&gt;
    my $num_args = @ARGV;&lt;br /&gt;
    if ($num_args &amp;gt;= 1) {$count = $ARGV[0];}&lt;br /&gt;
    if ($num_args &amp;gt;= 2) {$interval = $ARGV[1];}&lt;br /&gt;
    if ($num_args &amp;gt;= 3) {$output_file = $ARGV[2];}&lt;br /&gt;
    &lt;br /&gt;
    # check argument validity&lt;br /&gt;
    $0 =~ /^.*\\(.*)$/;&lt;br /&gt;
    my $script = $1;&lt;br /&gt;
    if ($count !~ /^\d+$/ or $interval !~ /^\d+$/) {die &amp;quot;Usage: $script &amp;lt;number of rounds&amp;gt; &amp;lt;seconds between rounds&amp;gt; &amp;lt;output.csv&amp;gt;\n&amp;quot;;}&lt;br /&gt;
    if (length($output_file) &amp;gt; 0 and $output_file !~ /\.csv$/) {$output_file .= &amp;quot;.csv&amp;quot;;}&lt;br /&gt;
    &lt;br /&gt;
    # main loop&lt;br /&gt;
    for my $i (1..$count)&lt;br /&gt;
    {&lt;br /&gt;
        sleep $interval unless $i == 1;&lt;br /&gt;
        print &amp;quot;Round $i\n&amp;quot;;&lt;br /&gt;
        my $host_num=0;&lt;br /&gt;
        foreach my $host (@hosts)&lt;br /&gt;
        {&lt;br /&gt;
            (my $ret, my $duration, my $ip) = $p-&amp;gt;ping($host);&lt;br /&gt;
            $ip =~ /(\d+)\.(\d+)\.(\d+)\.(\d+)/; &lt;br /&gt;
            if ($ret)&lt;br /&gt;
            {&lt;br /&gt;
                printf(&amp;quot;%*s [ip: %3s.%3s.%3s.%3s] is alive (%6.2f ms)\n&amp;quot;, $max_name_length, $host, $1, $2, $3, $4, $duration*1000);&lt;br /&gt;
                $data[$host_num][$i]=$duration*1000;&lt;br /&gt;
            }&lt;br /&gt;
            else&lt;br /&gt;
            {&lt;br /&gt;
                printf(&amp;quot;%*s [ip: %3s.%3s.%3s.%3s] is dead\n&amp;quot;, $max_name_length, $host, $1, $2, $3, $4);&lt;br /&gt;
            }&lt;br /&gt;
            $host_num++;&lt;br /&gt;
        }&lt;br /&gt;
        print &amp;quot;\n&amp;quot;;&lt;br /&gt;
    }&lt;br /&gt;
    $p-&amp;gt;close();&lt;br /&gt;
    &lt;br /&gt;
    # if output file name given&lt;br /&gt;
    if (length($output_file)&amp;gt;0)&lt;br /&gt;
    {&lt;br /&gt;
        # print output to file&lt;br /&gt;
        open FILE, &amp;quot;&amp;gt;$output_file&amp;quot; or die &amp;quot;$!\n&amp;quot;;&lt;br /&gt;
        &lt;br /&gt;
        # print column headers&lt;br /&gt;
        print FILE &amp;quot;Server\\Round&amp;quot;;&lt;br /&gt;
        for my $i (1..$count)&lt;br /&gt;
        {&lt;br /&gt;
            print FILE &amp;quot;, $i&amp;quot;;&lt;br /&gt;
        }&lt;br /&gt;
        print FILE &amp;quot;, Average\n&amp;quot;;&lt;br /&gt;
        &lt;br /&gt;
        # print data&lt;br /&gt;
        my $i = 0;&lt;br /&gt;
        foreach my $host (@hosts)&lt;br /&gt;
        {&lt;br /&gt;
            print FILE &amp;quot;$host&amp;quot;;&lt;br /&gt;
            my $sum = 0;&lt;br /&gt;
            for my $j (1..$count)&lt;br /&gt;
            {&lt;br /&gt;
                $sum += $data[$i][$j];&lt;br /&gt;
                printf FILE &amp;quot;, %8.4f&amp;quot;,$data[$i][$j];&lt;br /&gt;
            }&lt;br /&gt;
            printf FILE &amp;quot;, %8.4f\n&amp;quot;,$sum/$count;&lt;br /&gt;
            $i++;&lt;br /&gt;
        }&lt;br /&gt;
        &lt;br /&gt;
        close FILE;&lt;br /&gt;
        print &amp;quot;Data written to $output_file\n&amp;quot;;&lt;br /&gt;
    }&lt;br /&gt;
    &lt;br /&gt;
    # print summary to screen&lt;br /&gt;
    my $i = 0;&lt;br /&gt;
    printf(&amp;quot;%-*s Average (ms)\n&amp;quot;, $max_name_length, &amp;quot;Server&amp;quot;);&lt;br /&gt;
    foreach my $host (@hosts)&lt;br /&gt;
    {&lt;br /&gt;
        my $sum = 0;&lt;br /&gt;
        for my $j (1..$count)&lt;br /&gt;
        {&lt;br /&gt;
            $sum += $data[$i][$j];&lt;br /&gt;
        }&lt;br /&gt;
        printf(&amp;quot;%-*s %8.4f\n&amp;quot;, $max_name_length+1, $host, $sum/$count);&lt;br /&gt;
        $i++;&lt;br /&gt;
    }&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Output:&lt;br /&gt;
    Round 1&lt;br /&gt;
        atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.97 ms)&lt;br /&gt;
       atlanta2.voip.ms [ip:  72.  9.246.170] is alive ( 92.99 ms)&lt;br /&gt;
        chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.70 ms)&lt;br /&gt;
       chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 59.76 ms)&lt;br /&gt;
       chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.53 ms)&lt;br /&gt;
       chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 49.73 ms)&lt;br /&gt;
         dallas.voip.ms [ip:  74. 54. 54.178] is alive ( 94.99 ms)&lt;br /&gt;
         denver.voip.ms [ip: 173.248.161. 90] is alive ( 94.05 ms)&lt;br /&gt;
        denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.13 ms)&lt;br /&gt;
        houston.voip.ms [ip: 209. 62.  1.  2] is alive (102.87 ms)&lt;br /&gt;
     losangeles.voip.ms [ip:  96. 44.149.186] is alive ( 64.92 ms)&lt;br /&gt;
    losangeles2.voip.ms [ip:  96. 44.149.202] is alive ( 63.41 ms)&lt;br /&gt;
        newyork.voip.ms [ip:  74. 63. 41.218] is alive (131.75 ms)&lt;br /&gt;
       newyork2.voip.ms [ip: 107.  6. 67.236] is alive (120.64 ms)&lt;br /&gt;
       newyork3.voip.ms [ip: 107.  6. 67.237] is alive (120.49 ms)&lt;br /&gt;
       newyork4.voip.ms [ip: 107.  6. 67.238] is alive (111.43 ms)&lt;br /&gt;
        seattle.voip.ms [ip:  50. 23.160. 50] is alive ( 94.25 ms)&lt;br /&gt;
       seattle2.voip.ms [ip:  50. 23.160. 51] is alive ( 95.86 ms)&lt;br /&gt;
       seattle3.voip.ms [ip:  50. 23.160. 52] is alive ( 90.85 ms)&lt;br /&gt;
          tampa.voip.ms [ip:  68.233.226. 97] is alive (123.29 ms)&lt;br /&gt;
     washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.71 ms)&lt;br /&gt;
    washington2.voip.ms [ip: 208. 43.234.227] is alive (101.19 ms)&lt;br /&gt;
       montreal.voip.ms [ip:  67.205. 74.184] is alive ( 81.82 ms)&lt;br /&gt;
      montreal2.voip.ms [ip:  67.205. 74.187] is alive ( 86.13 ms)&lt;br /&gt;
      montreal3.voip.ms [ip:  72. 55.168. 18] is alive ( 77.09 ms)&lt;br /&gt;
      montreal4.voip.ms [ip:  67.205. 74.179] is alive ( 96.18 ms)&lt;br /&gt;
       toronto2.voip.ms [ip: 184. 75.215.114] is alive (103.70 ms)&lt;br /&gt;
       toronto3.voip.ms [ip: 184. 75.215.146] is alive (131.27 ms)&lt;br /&gt;
       toronto4.voip.ms [ip: 184. 75.213.210] is alive (125.13 ms)&lt;br /&gt;
        toronto.voip.ms [ip: 184. 75.215.106] is alive (103.26 ms)&lt;br /&gt;
         london.voip.ms [ip:   5. 77. 36.136] is alive (152.77 ms)&lt;br /&gt;
    &lt;br /&gt;
    Round 2&lt;br /&gt;
        atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.14 ms)&lt;br /&gt;
       atlanta2.voip.ms [ip:  72.  9.246.170] is alive ( 92.86 ms)&lt;br /&gt;
        chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 50.03 ms)&lt;br /&gt;
       chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 59.44 ms)&lt;br /&gt;
       chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.33 ms)&lt;br /&gt;
       chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 50.22 ms)&lt;br /&gt;
         dallas.voip.ms [ip:  74. 54. 54.178] is alive ( 95.58 ms)&lt;br /&gt;
         denver.voip.ms [ip: 173.248.161. 90] is alive ( 95.94 ms)&lt;br /&gt;
        denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.29 ms)&lt;br /&gt;
        houston.voip.ms [ip: 209. 62.  1.  2] is alive (102.73 ms)&lt;br /&gt;
     losangeles.voip.ms [ip:  96. 44.149.186] is alive ( 65.59 ms)&lt;br /&gt;
    losangeles2.voip.ms [ip:  96. 44.149.202] is alive ( 64.27 ms)&lt;br /&gt;
        newyork.voip.ms [ip:  74. 63. 41.218] is alive (112.74 ms)&lt;br /&gt;
       newyork2.voip.ms [ip: 107.  6. 67.236] is alive (121.22 ms)&lt;br /&gt;
       newyork3.voip.ms [ip: 107.  6. 67.237] is alive (121.34 ms)&lt;br /&gt;
       newyork4.voip.ms [ip: 107.  6. 67.238] is alive (110.75 ms)&lt;br /&gt;
        seattle.voip.ms [ip:  50. 23.160. 50] is alive ( 94.06 ms)&lt;br /&gt;
       seattle2.voip.ms [ip:  50. 23.160. 51] is alive ( 95.33 ms)&lt;br /&gt;
       seattle3.voip.ms [ip:  50. 23.160. 52] is alive ( 91.58 ms)&lt;br /&gt;
          tampa.voip.ms [ip:  68.233.226. 97] is alive (122.94 ms)&lt;br /&gt;
     washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.28 ms)&lt;br /&gt;
    washington2.voip.ms [ip: 208. 43.234.227] is alive (101.40 ms)&lt;br /&gt;
       montreal.voip.ms [ip:  67.205. 74.184] is alive ( 81.91 ms)&lt;br /&gt;
      montreal2.voip.ms [ip:  67.205. 74.187] is alive ( 85.64 ms)&lt;br /&gt;
      montreal3.voip.ms [ip:  72. 55.168. 18] is alive ( 75.15 ms)&lt;br /&gt;
      montreal4.voip.ms [ip:  67.205. 74.179] is alive ( 96.79 ms)&lt;br /&gt;
       toronto2.voip.ms [ip: 184. 75.215.114] is alive (103.10 ms)&lt;br /&gt;
       toronto3.voip.ms [ip: 184. 75.215.146] is alive (150.85 ms)&lt;br /&gt;
       toronto4.voip.ms [ip: 184. 75.213.210] is alive (138.40 ms)&lt;br /&gt;
        toronto.voip.ms [ip: 184. 75.215.106] is alive (103.45 ms)&lt;br /&gt;
         london.voip.ms [ip:   5. 77. 36.136] is alive (170.79 ms)&lt;br /&gt;
    &lt;br /&gt;
    Round 3&lt;br /&gt;
        atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 88.76 ms)&lt;br /&gt;
       atlanta2.voip.ms [ip:  72.  9.246.170] is alive ( 92.86 ms)&lt;br /&gt;
        chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.65 ms)&lt;br /&gt;
       chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 60.01 ms)&lt;br /&gt;
       chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.05 ms)&lt;br /&gt;
       chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 49.53 ms)&lt;br /&gt;
         dallas.voip.ms [ip:  74. 54. 54.178] is alive ( 95.82 ms)&lt;br /&gt;
         denver.voip.ms [ip: 173.248.161. 90] is alive ( 95.02 ms)&lt;br /&gt;
        denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.60 ms)&lt;br /&gt;
        houston.voip.ms [ip: 209. 62.  1.  2] is alive (103.35 ms)&lt;br /&gt;
     losangeles.voip.ms [ip:  96. 44.149.186] is alive ( 65.79 ms)&lt;br /&gt;
    losangeles2.voip.ms [ip:  96. 44.149.202] is alive ( 64.05 ms)&lt;br /&gt;
        newyork.voip.ms [ip:  74. 63. 41.218] is alive (113.01 ms)&lt;br /&gt;
       newyork2.voip.ms [ip: 107.  6. 67.236] is alive (121.41 ms)&lt;br /&gt;
       newyork3.voip.ms [ip: 107.  6. 67.237] is alive (122.23 ms)&lt;br /&gt;
       newyork4.voip.ms [ip: 107.  6. 67.238] is alive (110.62 ms)&lt;br /&gt;
        seattle.voip.ms [ip:  50. 23.160. 50] is alive ( 93.65 ms)&lt;br /&gt;
       seattle2.voip.ms [ip:  50. 23.160. 51] is alive ( 95.19 ms)&lt;br /&gt;
       seattle3.voip.ms [ip:  50. 23.160. 52] is alive ( 90.75 ms)&lt;br /&gt;
          tampa.voip.ms [ip:  68.233.226. 97] is alive (125.12 ms)&lt;br /&gt;
     washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.19 ms)&lt;br /&gt;
    washington2.voip.ms [ip: 208. 43.234.227] is alive (101.98 ms)&lt;br /&gt;
       montreal.voip.ms [ip:  67.205. 74.184] is alive ( 80.16 ms)&lt;br /&gt;
      montreal2.voip.ms [ip:  67.205. 74.187] is alive ( 87.16 ms)&lt;br /&gt;
      montreal3.voip.ms [ip:  72. 55.168. 18] is alive ( 76.54 ms)&lt;br /&gt;
      montreal4.voip.ms [ip:  67.205. 74.179] is alive ( 97.51 ms)&lt;br /&gt;
       toronto2.voip.ms [ip: 184. 75.215.114] is alive (104.18 ms)&lt;br /&gt;
       toronto3.voip.ms [ip: 184. 75.215.146] is alive (142.81 ms)&lt;br /&gt;
       toronto4.voip.ms [ip: 184. 75.213.210] is alive (138.95 ms)&lt;br /&gt;
        toronto.voip.ms [ip: 184. 75.215.106] is alive (103.78 ms)&lt;br /&gt;
         london.voip.ms [ip:   5. 77. 36.136] is alive (153.14 ms)&lt;br /&gt;
    &lt;br /&gt;
    Round 4&lt;br /&gt;
        atlanta.voip.ms [ip: 174. 34.146.162] is alive ( 89.19 ms)&lt;br /&gt;
       atlanta2.voip.ms [ip:  72.  9.246.170] is alive ( 92.98 ms)&lt;br /&gt;
        chicago.voip.ms [ip: 208.100. 39. 52] is alive ( 49.21 ms)&lt;br /&gt;
       chicago2.voip.ms [ip: 208.100. 39. 53] is alive ( 60.50 ms)&lt;br /&gt;
       chicago3.voip.ms [ip: 208.100. 39. 54] is alive ( 59.68 ms)&lt;br /&gt;
       chicago4.voip.ms [ip: 208.100. 39. 55] is alive ( 50.18 ms)&lt;br /&gt;
         dallas.voip.ms [ip:  74. 54. 54.178] is alive ( 93.93 ms)&lt;br /&gt;
         denver.voip.ms [ip: 173.248.161. 90] is alive ( 94.22 ms)&lt;br /&gt;
        denver2.voip.ms [ip: 173.248.159.210] is alive ( 85.10 ms)&lt;br /&gt;
        houston.voip.ms [ip: 209. 62.  1.  2] is alive (103.67 ms)&lt;br /&gt;
     losangeles.voip.ms [ip:  96. 44.149.186] is alive ( 65.58 ms)&lt;br /&gt;
    losangeles2.voip.ms [ip:  96. 44.149.202] is alive ( 63.60 ms)&lt;br /&gt;
        newyork.voip.ms [ip:  74. 63. 41.218] is alive (114.76 ms)&lt;br /&gt;
       newyork2.voip.ms [ip: 107.  6. 67.236] is alive (120.44 ms)&lt;br /&gt;
       newyork3.voip.ms [ip: 107.  6. 67.237] is alive (121.05 ms)&lt;br /&gt;
       newyork4.voip.ms [ip: 107.  6. 67.238] is alive (110.51 ms)&lt;br /&gt;
        seattle.voip.ms [ip:  50. 23.160. 50] is alive ( 94.04 ms)&lt;br /&gt;
       seattle2.voip.ms [ip:  50. 23.160. 51] is alive ( 96.92 ms)&lt;br /&gt;
       seattle3.voip.ms [ip:  50. 23.160. 52] is alive ( 91.23 ms)&lt;br /&gt;
          tampa.voip.ms [ip:  68.233.226. 97] is alive (123.28 ms)&lt;br /&gt;
     washington.voip.ms [ip: 208. 43.234.226] is alive ( 98.45 ms)&lt;br /&gt;
    washington2.voip.ms [ip: 208. 43.234.227] is alive (100.94 ms)&lt;br /&gt;
       montreal.voip.ms [ip:  67.205. 74.184] is alive ( 82.33 ms)&lt;br /&gt;
      montreal2.voip.ms [ip:  67.205. 74.187] is alive ( 85.02 ms)&lt;br /&gt;
      montreal3.voip.ms [ip:  72. 55.168. 18] is alive ( 76.85 ms)&lt;br /&gt;
      montreal4.voip.ms [ip:  67.205. 74.179] is alive ( 96.32 ms)&lt;br /&gt;
       toronto2.voip.ms [ip: 184. 75.215.114] is alive (104.22 ms)&lt;br /&gt;
       toronto3.voip.ms [ip: 184. 75.215.146] is alive (148.33 ms)&lt;br /&gt;
       toronto4.voip.ms [ip: 184. 75.213.210] is alive (141.61 ms)&lt;br /&gt;
        toronto.voip.ms [ip: 184. 75.215.106] is alive (105.91 ms)&lt;br /&gt;
         london.voip.ms [ip:   5. 77. 36.136] is alive (152.85 ms)&lt;br /&gt;
    &lt;br /&gt;
    Server              Average (ms)&lt;br /&gt;
    atlanta.voip.ms       88.7630&lt;br /&gt;
    atlanta2.voip.ms      92.9233&lt;br /&gt;
    chicago.voip.ms       49.6477&lt;br /&gt;
    chicago2.voip.ms      59.9305&lt;br /&gt;
    chicago3.voip.ms      59.3972&lt;br /&gt;
    chicago4.voip.ms      49.9152&lt;br /&gt;
    dallas.voip.ms        95.0790&lt;br /&gt;
    denver.voip.ms        94.8077&lt;br /&gt;
    denver2.voip.ms       85.2797&lt;br /&gt;
    houston.voip.ms      103.1562&lt;br /&gt;
    losangeles.voip.ms    65.4693&lt;br /&gt;
    losangeles2.voip.ms   63.8347&lt;br /&gt;
    newyork.voip.ms      118.0643&lt;br /&gt;
    newyork2.voip.ms     120.9265&lt;br /&gt;
    newyork3.voip.ms     121.2778&lt;br /&gt;
    newyork4.voip.ms     110.8275&lt;br /&gt;
    seattle.voip.ms       93.9993&lt;br /&gt;
    seattle2.voip.ms      95.8267&lt;br /&gt;
    seattle3.voip.ms      91.1035&lt;br /&gt;
    tampa.voip.ms        123.6570&lt;br /&gt;
    washington.voip.ms    98.4065&lt;br /&gt;
    washington2.voip.ms  101.3774&lt;br /&gt;
    montreal.voip.ms      81.5525&lt;br /&gt;
    montreal2.voip.ms     85.9863&lt;br /&gt;
    montreal3.voip.ms     76.4058&lt;br /&gt;
    montreal4.voip.ms     96.7013&lt;br /&gt;
    toronto2.voip.ms     103.7986&lt;br /&gt;
    toronto3.voip.ms     143.3156&lt;br /&gt;
    toronto4.voip.ms     136.0254&lt;br /&gt;
    toronto.voip.ms      104.1012&lt;br /&gt;
    london.voip.ms       157.3885&lt;br /&gt;
&lt;br /&gt;
=== Powershell ===&lt;br /&gt;
&lt;br /&gt;
 Dec 2017 - A bug in the code shown washington2.voip.ms as the best server, this was corrected.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;pre&amp;gt;&lt;br /&gt;
# Usage: Copy and paste the following code into a powershell window&lt;br /&gt;
# To run it from a command prompt, save this file with extension ps1. &lt;br /&gt;
# Then run Powershell.exe -file &amp;quot;pathtothisscript.ps1&amp;quot;&lt;br /&gt;
Clear-Variable best* -Scope Global #Clear the best* variables in case you run it more than once...&lt;br /&gt;
#Get the list of servers into an array&lt;br /&gt;
$Servers =      &lt;br /&gt;
@(&amp;quot;amsterdam.voip.ms&amp;quot;,&amp;quot;atlanta.voip.ms&amp;quot;,&amp;quot;atlanta2.voip.ms&amp;quot;,&amp;quot;chicago.voip.ms&amp;quot;,&amp;quot;chicago2.voip.ms&amp;quot;,&amp;quot;chicago3.voip.ms&amp;quot;,&lt;br /&gt;
&amp;quot;chicago4.voip.ms&amp;quot;,&amp;quot;dallas.voip.ms&amp;quot;,&amp;quot;dallas2.voip.ms&amp;quot;,&amp;quot;denver.voip.ms&amp;quot;,&amp;quot;denver2.voip.ms&amp;quot;,&amp;quot;houston.voip.ms&amp;quot;,&lt;br /&gt;
&amp;quot;houston2.voip.ms&amp;quot;,&amp;quot;london.voip.ms&amp;quot;,&amp;quot;losangeles.voip.ms&amp;quot;,&amp;quot;losangeles2.voip.ms&amp;quot;,&amp;quot;montreal.voip.ms&amp;quot;,&lt;br /&gt;
&amp;quot;montreal2.voip.ms&amp;quot;,&amp;quot;montreal3.voip.ms&amp;quot;,&amp;quot;montreal4.voip.ms&amp;quot;,&amp;quot;montreal5.voip.ms&amp;quot;,&amp;quot;montreal6.voip.ms&amp;quot;,&amp;quot;montreal7.voip.ms&amp;quot;,&lt;br /&gt;
&amp;quot;montreal8.voip.ms&amp;quot;,&amp;quot;newyork.voip.ms&amp;quot;,&amp;quot;newyork2.voip.ms&amp;quot;,&amp;quot;newyork3.voip.ms&amp;quot;,&amp;quot;newyork4.voip.ms&amp;quot;,&amp;quot;newyork5.voip.ms&amp;quot;,&lt;br /&gt;
&amp;quot;newyork6.voip.ms&amp;quot;,&amp;quot;newyork7.voip.ms&amp;quot;,&amp;quot;newyork8.voip.ms&amp;quot;,&amp;quot;paris.voip.ms&amp;quot;,&amp;quot;sanjose.voip.ms&amp;quot;,&amp;quot;sanjose2.voip.ms&amp;quot;,&lt;br /&gt;
&amp;quot;seattle.voip.ms&amp;quot;,&amp;quot;seattle2.voip.ms&amp;quot;,&amp;quot;seattle3.voip.ms&amp;quot;,&amp;quot;tampa.voip.ms&amp;quot;,&amp;quot;tampa2.voip.ms&amp;quot;,&amp;quot;toronto.voip.ms&amp;quot;,&lt;br /&gt;
&amp;quot;toronto2.voip.ms&amp;quot;,&amp;quot;toronto3.voip.ms&amp;quot;,&amp;quot;toronto4.voip.ms&amp;quot;,&amp;quot;toronto5.voip.ms&amp;quot;,&amp;quot;toronto6.voip.ms&amp;quot;,&amp;quot;toronto7.voip.ms&amp;quot;,&lt;br /&gt;
&amp;quot;toronto8.voip.ms&amp;quot;,&amp;quot;vancouver.voip.ms&amp;quot;,&amp;quot;vancouver2.voip.ms&amp;quot;,&amp;quot;washington.voip.ms&amp;quot;,&amp;quot;washington2.voip.ms&amp;quot;)&lt;br /&gt;
$k = 0 #Counting variable so we know what server number we are testing&lt;br /&gt;
#num of servers to test&lt;br /&gt;
$servercount = $servers.length &lt;br /&gt;
#Do the following code for each server in our array&lt;br /&gt;
ForEach($server in $servers)&lt;br /&gt;
{  &lt;br /&gt;
  #Add one to the counting variable....we are on server #1...then server 2, then server 3 etc...&lt;br /&gt;
  $k++&lt;br /&gt;
  #Update the progress bar                    &lt;br /&gt;
  Write-Progress -Activity &amp;quot;Testing Server: ${server}&amp;quot; -status &amp;quot;Testing Server $k out of $servercount&amp;quot; -percentComplete ($k / $servercount*100) &lt;br /&gt;
  #Counting variable for number of times we tried to ping a given server&lt;br /&gt;
  $i = 0&lt;br /&gt;
  Do{&lt;br /&gt;
     #assume a failure&lt;br /&gt;
     $pingsuccess = $false &lt;br /&gt;
     $i++ #Add one to the counting variable.....1st try....2nd try....3rd try etc...&lt;br /&gt;
     Try{&lt;br /&gt;
         #Try to ping&lt;br /&gt;
         $currentping = (test-connection $server -count 1 -ErrorAction Stop).responsetime &lt;br /&gt;
         #If success full, set success variable&lt;br /&gt;
         $pingsuccess = $true&lt;br /&gt;
     }&lt;br /&gt;
     #Catch the failure and set the success variable to false&lt;br /&gt;
     Catch {&lt;br /&gt;
      $pingsuccess = $false &lt;br /&gt;
      }     &lt;br /&gt;
  }&lt;br /&gt;
  #Try everything between Do and While up to 5 times, or while $pingsuccess is not true&lt;br /&gt;
  While($pingsuccess -eq $false -and $i -le 5) &lt;br /&gt;
  #Compare the last ping test with the best known ping test....if there is no known best ping test, assume this one is the best $bestping = $currentping &lt;br /&gt;
  If($pingsuccess -and ($currentping -lt $bestping -or (!($bestping)))){ &lt;br /&gt;
  #If this is the best ping...save it&lt;br /&gt;
        $bestserver = $server    #Save the best server&lt;br /&gt;
        $bestping = $currentping #Save the best ping results&lt;br /&gt;
  }&lt;br /&gt;
  write-host &amp;quot;tested: $server at $currentping ms after $i attempts&amp;quot; #write the results of the test for this server&lt;br /&gt;
}&lt;br /&gt;
write-host &amp;quot;`r`n The server with the best ping is: $bestserver at $bestping ms`r`n&amp;quot; #write the end result&lt;br /&gt;
Pause&lt;br /&gt;
&amp;lt;/pre&amp;gt;&lt;br /&gt;
&lt;br /&gt;
=== Linux Shell Script ===&lt;br /&gt;
Pings several voip.ms servers&lt;br /&gt;
&lt;br /&gt;
   #!/bin/sh&lt;br /&gt;
   # Ping several servers and display Latency, Jitter and Packet Loss &lt;br /&gt;
   #&lt;br /&gt;
   # First, create a text file with all servers you want to ping - one host name per line. &lt;br /&gt;
   # The list of voip.ms servers is available at http://wiki.voip.ms/article/Choosing_Server&lt;br /&gt;
   myHF=&amp;quot;voip_ping_hosts.txt&amp;quot;&lt;br /&gt;
   # Sample file:&lt;br /&gt;
   #    toronto.voip.ms&lt;br /&gt;
   #    montreal.voip.ms&lt;br /&gt;
   #    seattle.voip.ms&lt;br /&gt;
   #    chicago.voip.ms&lt;br /&gt;
   #    newyork.voip.ms&lt;br /&gt;
   #&lt;br /&gt;
   echo &amp;quot;============================================&amp;quot;&lt;br /&gt;
   printf &amp;quot;%-20s %7s %8s %6s\n&amp;quot; &amp;quot;VoIP Server&amp;quot; &amp;quot;Latency&amp;quot; &amp;quot;Jitter&amp;quot; &amp;quot;Loss&amp;quot;&lt;br /&gt;
   echo &amp;quot;============================================&amp;quot;&lt;br /&gt;
   cat ${myHF} |\&lt;br /&gt;
   while read myLn&lt;br /&gt;
   do&lt;br /&gt;
      ping -c 3 -i 5 -q $myLn |\&lt;br /&gt;
      awk '/^PING / {myH=$2}&lt;br /&gt;
           /packet loss/ {myPL=$6}&lt;br /&gt;
           /min\/avg\/max/ {&lt;br /&gt;
              split($4,myS,&amp;quot;/&amp;quot;)&lt;br /&gt;
              printf( &amp;quot;%-20s    %3.1f    %1.3f   %4s\n&amp;quot;, myH, myS[2], myS[4], myPL)&lt;br /&gt;
          }'&lt;br /&gt;
   done&lt;br /&gt;
   echo &amp;quot;============================================&amp;quot;&lt;br /&gt;
&lt;br /&gt;
Output:&lt;br /&gt;
&lt;br /&gt;
   ============================================&lt;br /&gt;
   VoIP Server          Latency   Jitter   Loss&lt;br /&gt;
   ============================================&lt;br /&gt;
   toronto.voip.ms         68.3    0.439     0%&lt;br /&gt;
   montreal.voip.ms        89.6    0.197     0%&lt;br /&gt;
   seattle.voip.ms         71.2    0.387     0%&lt;br /&gt;
   chicago.voip.ms         71.6    0.084     0%&lt;br /&gt;
   newyork.voip.ms         79.1    0.411     0%&lt;br /&gt;
   ============================================&lt;br /&gt;
&lt;br /&gt;
= Latency and its importance =&lt;br /&gt;
&lt;br /&gt;
Latency is very important for Voip, this will determine the time that will take for the data package transmission to reach the destination. A high latency will lead to a delay and echoes in the communication.&lt;br /&gt;
&lt;br /&gt;
Latency is measured in milliseconds (ms) For example: a latency of 150ms is barely noticeable, thus acceptable. Higher than that, quality starts to suffer. When it gets higher than 300 ms, it becomes unacceptable.&lt;br /&gt;
&lt;br /&gt;
[[category:guides]]&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/File:ChooseServerImg.png</id>
		<title>File:ChooseServerImg.png</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/File:ChooseServerImg.png"/>
				<updated>2021-07-16T15:52:46Z</updated>
		
		<summary type="html">&lt;p&gt;William: uploaded a new version of &amp;amp;quot;File:ChooseServerImg.png&amp;amp;quot;&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;New Chose Server Image.&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/Audio_Conferencing</id>
		<title>Audio Conferencing</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/Audio_Conferencing"/>
				<updated>2018-10-24T15:18:39Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;__TOC__&lt;br /&gt;
&lt;br /&gt;
The Audio Conferencing feature allows several people to dial into your DID number to be bridged together in the same conversation.&amp;lt;br&amp;gt;&lt;br /&gt;
Audio Conferencing provides the host with a dashboard monitor that allows organizers to monitor and control different aspects of the conference call in real time.&lt;br /&gt;
Press the '''“Start Recording”''' button to get a recording of your conference call that you can listen and download after for any purposes.&amp;lt;br&amp;gt;&lt;br /&gt;
Start using the Audio Conferencing feature by clicking the '''“Add Conference”''' button to create an entry. Once your conference entry is ready you can route your DID number to it from the Manage DIDs &amp;gt; Edit interface to start receiving calls into your conference.&lt;br /&gt;
&lt;br /&gt;
 '''Note: Conference feature is only available to DID Numbers in Per Minute plan.'''&lt;br /&gt;
&lt;br /&gt;
'''Calls entering the Audio Conferencing are billed each one independently according to the regular per minute rate of the DID number for the whole duration that participant remained in the conference.'''&lt;br /&gt;
&lt;br /&gt;
[[File:Conf1.png]]&lt;br /&gt;
&lt;br /&gt;
Click on the '''“Add Conference”''' button to start creating your Audio Conferencing entry.&lt;br /&gt;
&lt;br /&gt;
[[File:Addconf.png]]&lt;br /&gt;
&lt;br /&gt;
*'''Name:''' This is the unique name of your conference. (e.g. Sales Meeting)&lt;br /&gt;
*'''Description:''' This can be used as a note or description to easily identify your conference.&lt;br /&gt;
*'''Maximum Participants:''' Specifies the maximum amount of callers that can participate in this conference. Leave empty for no limit. Once the limit is reached, the conference will be locked until a person leaves. Admin-level users are exempt from this limit and will still be able to access to join a locked conference.&lt;br /&gt;
*'''Language:''' Language for system messages, such as “Invalid Option”.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br&amp;gt;&lt;br /&gt;
== Member Profile Basic Options ==&lt;br /&gt;
[[File:Confbasic.png]]&lt;br /&gt;
&lt;br /&gt;
*'''Name:''' This is the unique name of your Member Profile.&lt;br /&gt;
*'''Description:''' This can be used as a note or description to easily identify your Member Profile.&lt;br /&gt;
*'''PIN:''' Sets if the user must enter a PIN before joining the conference. The user will be prompted for the PIN.  This also serves for selecting the member profile when the user joins the conference.&lt;br /&gt;
&lt;br /&gt;
 '''Note: The usage of the pound key after the PIN number to join the conference can be included&lt;br /&gt;
 or excluded depending on your configuration, for instance, if all the members to the conference&lt;br /&gt;
 are granted with a PIN code with the same amount of digits, they won't be requested to dial pound&lt;br /&gt;
 key, however if one of the members does have a larger PIN code, other members will require to dial &lt;br /&gt;
pound key to access the conference.'''&lt;br /&gt;
&lt;br /&gt;
*'''Admin:''' Sets if the user is an Admin or not.&lt;br /&gt;
*'''Start Muted:''' Sets if the user should start out muted after entering the conference.&lt;br /&gt;
*'''Announce Join Leave:''' When enabled, this option prompts the user for his name while entering the conference. After the name is recorded, it will be played as the user enters and exits the conference.&lt;br /&gt;
*'''Announce User Count:''' Sets if the number of users in the conference should be announced to the caller as he joins.&lt;br /&gt;
*'''Announce Only User:''' Sets if the '''“only user”''' announcement should be played when a caller enters an empty conference.&lt;br /&gt;
&amp;lt;br&amp;gt;&lt;br /&gt;
== Member Profile Advanced Options ==&lt;br /&gt;
[[File:Confadvanced.png]]&lt;br /&gt;
&lt;br /&gt;
*'''MOH When Empty:''' Sets whether music on hold (MOH) should be played when only one person is in the conference.&lt;br /&gt;
*'''Quiet:''' When set to '''&amp;quot;YES&amp;quot;''', enter/leave prompts and user introductions are not played.&lt;br /&gt;
*'''Announcement:''' If set, this recording will be heard only by the user as he joins the conference.&lt;br /&gt;
*'''Drop Silence:''' The system will drop what is detected as silence from entering into the conference. Enabling this option will drastically improve performance and help remove the buildup of background noise from the call. This option is highly recommended for large conferences, due to its performance improvements.&lt;br /&gt;
*'''Talking Threshold:''' The time, in milliseconds, that a users needs to be sending sound or voice before the system can consider them to be talking. '''Recommended value is 160 MS.'''&lt;br /&gt;
&lt;br /&gt;
This setting affects different options:&lt;br /&gt;
&lt;br /&gt;
''1) Audio is only mixed out of a user's incoming audio stream if talking is detected. If this value is set too loose, the users will hear themselves briefly each time they begin talking until enough time has passed for the system to establish that they are in fact talking.''&amp;lt;br&amp;gt;&lt;br /&gt;
''2) This value determines when talking has begun, which causes events to trigger in the conference dashboard. If this value is set too tight, events may be falsely triggered by the background noise of the caller.''&amp;lt;br&amp;gt;&lt;br /&gt;
''3) The '''&amp;quot;Drop Silence&amp;quot;''' option depends on this value to determine when the user's audio should be mixed into the conference after periods of silence. If this value is too loose, the beginning of a user's speech will get cut off as they transition from silence to talking.''&lt;br /&gt;
*'''Silence Threshold:''' The time, in milliseconds, that silence needs to be present in the user’s sound stream before the system can consider it to be in fact silent and close the audio.&lt;br /&gt;
The best way to use this option is to set it slightly above the maximum amount of milliseconds of silence a user may generate during natural speech, i.e. the regular pauses while speaking.&lt;br /&gt;
'''Recommended value is 2500 MS'''.&lt;br /&gt;
&lt;br /&gt;
This setting affects different options:&amp;lt;br&amp;gt;&lt;br /&gt;
''1) This value determines when the user has stopped talking after a period of talking. If this value is set too low, events in the conference dashboard indicating that the user has stopped talking may get falsely sent out when the user briefly pauses during mid sentence.''&amp;lt;br&amp;gt;&lt;br /&gt;
''2) The '''&amp;quot;Drop Silence&amp;quot;''' option depends on this value to determine when the user's audio should begin to be dropped from the conference, after the user stops talking. If this value is set too low, the user's audio stream may sound choppy to other participants.''&lt;br /&gt;
*'''Talk Detection:''' If set to '''YES''', the conference dashboard will display a notification when a participant starts and stops talking.&lt;br /&gt;
*'''Jitter Buffer:''' When set to '''YES''',  the system will place a jitter buffer on the caller's audio stream before any audio mixing is performed. This option is highly recommended for maximum voice quality, but will add a slight delay to the audio and slightly impact system’s performance.&lt;br /&gt;
&amp;lt;br&amp;gt;&lt;br /&gt;
== Conference Dashboard (BETA)==&lt;br /&gt;
&lt;br /&gt;
The dashboard will allow the user to have a graphical interface of the participants in the conference call, providing additional options for the admins. The Dashboard will also offer the '''&amp;quot;Start Recording&amp;quot;''' option, making it possible to record the call by the whole duration the call recording service was active. '''Call recording service has a rate of $0.0025 per minute''', for the duration of the recording. The per minute charges will appear in the CDR.&lt;br /&gt;
&lt;br /&gt;
[[File:Dash1.png]]&lt;br /&gt;
&lt;br /&gt;
The '''&amp;quot;Downloads&amp;quot;''' button opens a screen that allows the user to download and hear the call recordings directly from the user interface.&lt;br /&gt;
&lt;br /&gt;
[[File:Confdownload.png]]&lt;br /&gt;
&lt;br /&gt;
The option to send the call recording over email is also available. Simply input the target email address and click '''&amp;quot;Send Email&amp;quot;''' button.&lt;br /&gt;
&lt;br /&gt;
[[File:Sendemail.png]]&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
== Conference Recordings ==&lt;br /&gt;
[[File:Confrec.png]]&lt;br /&gt;
*'''Join:''' The recording played when a user joins, typically some kind of beep sound&lt;br /&gt;
*'''Leave:''' The recording played when a user leaves, typically some kind of beep sound&lt;br /&gt;
*'''Has joined:''' The recording played as a user intro, e.g. &amp;quot;xxxx has joined the conference.&amp;quot;&lt;br /&gt;
*'''Has left:''' The recording played as a user parts the conference, e.g. &amp;quot;xxxx has left the conference.&amp;quot;&lt;br /&gt;
*'''Kicked:''' The recording played to a user who has been kicked from the conference.&lt;br /&gt;
*'''Muted:''' The recording played to a user when the mute option is toggled on.&lt;br /&gt;
*'''Unmuted:''' The recording played to a user when the mute option is toggled off.&lt;br /&gt;
*'''Only person:''' The recording played when a user is the only person in the conference.&lt;br /&gt;
*'''Only one:''' The recording played to a user when there is only one other person in the conference.&lt;br /&gt;
*'''There are:''' The recording played when announcing how many users there are in a conference.&lt;br /&gt;
*'''Participants muted:''' The recording played when all non-admin participants are muted.&lt;br /&gt;
*'''Other in party:''' The recording used in conjunction with the &amp;quot;There are&amp;quot; option, used like &amp;quot;There are&amp;quot; (number of participants) &amp;quot;Other in party”&lt;br /&gt;
*'''Place into conference:''' The recording played when someone is placed into a conference, after waiting for a marked user.&lt;br /&gt;
*'''Get PIN:''' The recording played when prompting for a conference PIN.&lt;br /&gt;
*'''Invalid PIN:''' The recording played when an invalid PIN is entered too many (3) times.&lt;br /&gt;
*'''Locked:''' The recording played to a user trying to join a locked conference.&lt;br /&gt;
*'''Locked now:''' The recording played to an Admin-level user after toggling the conference to locked mode&lt;br /&gt;
*'''Unlocked now:''' The recording played to an Admin-level user after toggling the conference to unlocked mode.&lt;br /&gt;
*'''Error menu:''' The recording played when an invalid menu option is entered.&lt;br /&gt;
*'''Participants unmuted:''' The recording played when all non-admin participants are unmuted.&lt;br /&gt;
&lt;br /&gt;
&amp;lt;br&amp;gt;&lt;br /&gt;
== Additional Phone Options ==&lt;br /&gt;
&lt;br /&gt;
The following are additional advanced options native of asterisk.&lt;br /&gt;
&lt;br /&gt;
For a Regular user:&lt;br /&gt;
 *=Plays the options menu&lt;br /&gt;
 1=toggle_mute&lt;br /&gt;
 4=decrease_listening_volume&lt;br /&gt;
 6=increase_listening_volume&lt;br /&gt;
 7=decrease_talking_volume&lt;br /&gt;
 8=leave_conference&lt;br /&gt;
 9=increase_talking_volume&lt;br /&gt;
&lt;br /&gt;
For an Admin user:&lt;br /&gt;
 *=Plays the options menu (conf-adminmenu)&lt;br /&gt;
 1=toggle_mute&lt;br /&gt;
 2=admin_toggle_conference_lock  ; only applied to admin users&lt;br /&gt;
 3=admin_kick_last        ; only applied to admin users&lt;br /&gt;
 4=decrease_listening_volume&lt;br /&gt;
 6=increase_listening_volume&lt;br /&gt;
 7=decrease_talking_volume&lt;br /&gt;
 9=increase_talking_volume&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/Extra_Services_Costs</id>
		<title>Extra Services Costs</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/Extra_Services_Costs"/>
				<updated>2017-06-23T16:16:00Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;Although most of the services at VoIP.ms are free there are some services that cost extra, beyond the Basic [[Service Cost]]s, either by a one time only, per call or monthly fee.&lt;br /&gt;
&lt;br /&gt;
On this page you will see these services, what to do to enable or use them and how much they cost.&lt;br /&gt;
&lt;br /&gt;
===Porting a Number / DID into VoIP.ms Service===&lt;br /&gt;
There is a one time fee of '''$8.50* USD''' per Canadian number in order to port it to our network (this may be lower if there is a promotion running), the process can take from 1-4 weeks for Canadian     &lt;br /&gt;
Numbers and 2-4 weeks for US and Toll Free Numbers.&lt;br /&gt;
&lt;br /&gt;
You can start the Porting Procedure from the [https://www.voip.ms/m/didporting.php DID Portability] page.&lt;br /&gt;
&lt;br /&gt;
Please see the [[Porting a Number]] Page to get full details on Porting a Number.&lt;br /&gt;
&lt;br /&gt;
(*Holiday Exchange Rate Special will be available for a Limited Time.)&lt;br /&gt;
&lt;br /&gt;
===Porting Out a Number / DID from VoIP.ms Service===&lt;br /&gt;
At VoIP.ms sole discretion, you may incur a port away fee for any DID number(s) leaving our network as this is a pass-through charge from VoIP.ms carrier(s) or if you are caught using VoIP.ms as a DID Store to purchase DIDs and port them out immediately VoIP.ms reserves the right to charge you a porting out fee.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
===CNAM Initiation and Updates===&lt;br /&gt;
VoIP.ms cannot provide this service to all numbers so please contact support@voip.ms to make sure they can for your number. &lt;br /&gt;
&lt;br /&gt;
This process can take up to 15 business days and you should always test the CNAM with calling a landline.&lt;br /&gt;
&lt;br /&gt;
CallerID Name (CNAM) is an string of 15 alphanumeric characters that are associated with a given CallerID number.&lt;br /&gt;
&lt;br /&gt;
The sample below is a Caller ID, that includes both Caller ID name and Caller ID number, commonly abbreviated as CID and CNAM among other variations. &lt;br /&gt;
&lt;br /&gt;
It is not possible to set any Caller ID Name from the VoIP.ms portal. &lt;br /&gt;
&lt;br /&gt;
&amp;quot;John Smith&amp;quot; &amp;lt;9145551234&amp;gt;&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''In USA:'''&lt;br /&gt;
:There is a '''$10 fee''' to update or start the CNAM process, to get the name or update sent to the LIDB/CNAM Database. &lt;br /&gt;
&lt;br /&gt;
:All CNAM queries originate from this database or an up to date copy of it. Most of VoIP.ms carriers support this process. &lt;br /&gt;
&lt;br /&gt;
:Inbound CNAM [[Caller ID]] Name Lookup - '''0.8 cent''' (8/10 of a cent) per query. &lt;br /&gt;
&lt;br /&gt;
:CNAM Lookup will be charged if the call matches a North American CallerID (NPANXXXXXX) and whether the call is answered or not. You are not charged for a CNAM Lookup if the number is in your Customer Portal's [[Phone book]]&lt;br /&gt;
&lt;br /&gt;
'''In Canada:''' &lt;br /&gt;
:CallerID Name will pass on Outbound / Premium rate. VoIP.ms can also request the number to be submitted to the white pages (411), this will usually lead to the number getting a CNAM Entry to the LIDB/CNAM Database. However, the easiest way is to simply use the premium route and configure a name in the ATA Device, PBX or Softphone.&lt;br /&gt;
&lt;br /&gt;
VoIP.ms does not make any profit from this, they are just passing on the charges as they are charged for this service.&lt;br /&gt;
&lt;br /&gt;
On all requests, whether to start or to update it, VoIP.ms will need the following information in an email sent to support@voip.ms:&lt;br /&gt;
&lt;br /&gt;
:DID Number(s):&lt;br /&gt;
:CNAM:&lt;br /&gt;
:(Up to 15 alphanumeric characters, a-z and 0-9 allowed)&lt;br /&gt;
:Name:&lt;br /&gt;
:Address:    (No P.O. Boxes Please)&lt;br /&gt;
:City:&lt;br /&gt;
:State/Province:&lt;br /&gt;
:Zip/Postal Code:&lt;br /&gt;
:Contact Number:&lt;br /&gt;
&lt;br /&gt;
===E911 / 911===&lt;br /&gt;
Use of the E911 Service costs a one time fee of '''$1.50''' on activation and a regulatory monthly fee of '''$1.50''' per DID number activated per month. VoIP.MS does not make any profit on this charge, it is simply what must be paid to provide this service.&lt;br /&gt;
&lt;br /&gt;
Please note you can enable this service only for Canadian or US numbers (including USA or Canadian toll free numbers). You just have to click on the Apply button ([https://www.voip.ms/m/me911.php Click Here]) in order to start the process to enable the service for this number. &lt;br /&gt;
&lt;br /&gt;
If at any time you need to change your address, you just need to select the &amp;quot;Modify&amp;quot; option, then click on Apply button. After this you will be able to change your e911 information and after the information has been approved, you will be confirmed by email and the e911 service will be updated.&lt;br /&gt;
&lt;br /&gt;
Please read the [[E911|E911/911]] complete information page.&lt;br /&gt;
&lt;br /&gt;
===411 / Directory Assistance===&lt;br /&gt;
4-1-1: '''$0.99''' per call. &lt;br /&gt;
:This must be activated by the customer in [https://www.voip.ms/m/settings.php Account Settings] then under Account Restrictions Tab. &lt;br /&gt;
:Please note that XXX-555-1212 has been disabled in our system and you will not be able to call it.&lt;br /&gt;
&lt;br /&gt;
===Directory Listing / White Pages / 411 Listing===&lt;br /&gt;
VoIP.ms cannot provide this service to all numbers so please contact support@voip.ms to make sure they can for your number. &lt;br /&gt;
&lt;br /&gt;
''For Canadians:'' &lt;br /&gt;
:If your DID number can be listed in White Page Directory please also note that this process can take from 15 to 60 business days. &lt;br /&gt;
:There is a '''$4.50 USD''' setup fee (one time only). You will be charged the setup fee again if you have to update your entry in the future.&lt;br /&gt;
&lt;br /&gt;
''For USA Customers:''&lt;br /&gt;
:If your DID number can be listed in the White Page Directory please also note that this process can take from 15 to 60 business days. &lt;br /&gt;
:There is a '''$5.00 USD''' setup fee (one time only) and an additional '''$1.49 USD''' charge per month. You will be charged the setup fee again if you have to update your entry in the future.&lt;br /&gt;
&lt;br /&gt;
VoIP.ms does not make any profit from this, they are just passing on the charges as they are charged for this service.&lt;br /&gt;
&lt;br /&gt;
On ALL Requests, whether to start or to update, VoIP.ms will need the following information in an email to support@voip.ms:&lt;br /&gt;
&lt;br /&gt;
: Business Name: (if applicable)&lt;br /&gt;
: First Name: &lt;br /&gt;
: Last Name:&lt;br /&gt;
: Address: &lt;br /&gt;
: City:&lt;br /&gt;
: State: &lt;br /&gt;
: Zip Code: &lt;br /&gt;
: Email:&lt;br /&gt;
: Contact Number:&lt;br /&gt;
&lt;br /&gt;
===SMS Messages===&lt;br /&gt;
Short Message Service ([[SMS]]) allows you to send and receive messages with your DID Number (Canada and US DID numbers only) for '''1 cent per message''' receiving and sending (once this feature is no longer in BETA). SMS is not currently available on [[toll-free number]] or [[Order a DID Number#Ordering iNums|iNum]] DIDs. Inbound SMS may be forwarded to an e-mail address or mobile phone. A [https://www.voip.ms/images/telephone.png handset icon] is displayed on the [https://www.voip.ms/m/managedid.php manage DID number] page to indicate each SMS-capable DID.&lt;br /&gt;
&lt;br /&gt;
This feature is for regular customer usage. No automation, telemarketing, bulk sending or receiving will be allowed. For further information, please see [[SMS]].&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/Account_Settings</id>
		<title>Account Settings</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/Account_Settings"/>
				<updated>2017-02-14T21:35:42Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;The Account Settings page is one of the most important pages in the Customer Portal, it will help you out as its name points out, to set all the proper ways to use your account, and these settings will affect the global settings on your account, from Device type to routing and even security of your account. &lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
__TOC__&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
== Account Routing ==&lt;br /&gt;
&lt;br /&gt;
These settings define the routes the system will use when you place calls to either US, Canada, Toll Free or International Numbers.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accsettings1.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Canada Routing''': You can choose between the [[Value vs Premium|'''Value''' or '''Premium''']] Route. If you are looking for the best wholesale price without any volume commitment you can use the '''Value''' route, however the CallerID number is not always guaranteed to work in this route.&lt;br /&gt;
The '''Premium''' route has a better level of quality, at a price that is a little higher than the value option. The CallerID Number is guaranteed to work for this route. Please note that the '''Canada Route''' can also be set independently per sub account, under the [[Sub Accounts]] settings. &lt;br /&gt;
&lt;br /&gt;
  Please note the US route is only available on the Premium route with a rate of $0.009 per minute.&lt;br /&gt;
&lt;br /&gt;
'''International Routing''': You can choose also the [[Value vs Premium|'''Value''' or '''Premium''']] route. Both routes are reliable, however you're going to find better quality if you're using the '''Premium''' route. The CallerID Number is not guaranteed to work for international calls even if you use the '''Premium''' route.&lt;br /&gt;
&lt;br /&gt;
'''Toll Free Routing''': This define the route the system will use when you place a Toll-Free call to Canada or USA. You can choose between the '''Value''' route (free to use) or the '''Premium''' (at 1.06 cent per minute). As the other settings, the '''Premium''' route give you better quality and the CallerID Number is guaranteed to work.&lt;br /&gt;
&lt;br /&gt;
'''Toll Free Carrier''': This setting defines the type of carrier that will be used when you dial a Toll-Free number. You can choose between '''Server Default''', '''American Carrier''' or '''Canadian Carrier'''. &lt;br /&gt;
&lt;br /&gt;
 '''Server Default:''' The Carrier for the termination of your toll-free call will be chosen depending on the geographical location of the server &lt;br /&gt;
   you are registering or connecting to. American Servers will dial through an American Carrier and Canadian servers will use a Canadian Toll-Free carrier.&lt;br /&gt;
 '''American Carrier:''' Your Toll-Free calls will always go through an American Carrier, even if you use a Canadian server. &lt;br /&gt;
 '''Canadian Carrier:''' Your Toll-Free calls will always go through a Canadian Carrier, even if you use an American server.&lt;br /&gt;
&lt;br /&gt;
For example, this feature can be useful for west based Canadian customers who wish to use the Seattle server but have their toll-free termination calls go through a Canadian carrier.&lt;br /&gt;
&lt;br /&gt;
 '''Note''': Please refer to [[Service Cost]] for information about the rates of the outgoing calls.&lt;br /&gt;
&lt;br /&gt;
== Account Restrictions ==&lt;br /&gt;
&lt;br /&gt;
These settings define the restrictions the system will use when you place calls to either USA, Canada or International Numbers.&lt;br /&gt;
&lt;br /&gt;
[[File:Account Restrictions.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Allow 411 dialing''': When enabled, you can call 411 directory assistance at the cost of $0.99 per call. &lt;br /&gt;
&lt;br /&gt;
 '''Note''': XXX-555-1212 is disabled within VoIP.ms. The only way to reach directory assistance is by dialing 411.&lt;br /&gt;
&lt;br /&gt;
'''Allow International Calls''': When Disabled, calls placed to destinations outside USA/Canada will be automatically rejected. If you are using [[Sub Accounts]] you need to setup this directly in the sub account.&lt;br /&gt;
&lt;br /&gt;
 '''Note''': This option is set to &amp;quot;No&amp;quot; by default for New accounts.&lt;br /&gt;
&lt;br /&gt;
'''Max. Call Time for US48/Canadian Calls''': With these setting you can choose how long the call will last when you call to US or Canada Numbers. If a call to a US or Canada Number exceeds the maximum time you have set in this option, it will be automatically hung up.&lt;br /&gt;
&lt;br /&gt;
'''Max. Call Time for International Calls''': Works exactly as the '''Max. Call Time for US48/Canadian Calls''' setting. If a call to an International Number exceeds the maximum time set, it will be automatically hung up.&lt;br /&gt;
&lt;br /&gt;
'''International Amount Restriction''': With this option you can set the maximum amount per minute for calls to international destinations. If the Per minute rate for an International destination exceeds the value you have set in this option, the call will not be connected.&lt;br /&gt;
&lt;br /&gt;
For example, let say you can set this option to $0.150 and you want to call a UK mobile number with area code 4470, the call will not go through because the per minute rate for this area code is $0.3048.&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' This settings depends on the International Routing you have choose.&lt;br /&gt;
&lt;br /&gt;
'''Allow Calls to Countries''': Here you can select Regions or Specific Countries to allow outgoing calls from your account. &lt;br /&gt;
&lt;br /&gt;
In order to allow calls to a specific country you can click the name of the region to expand the countries within it and select the desired countries only. You can select '''Select All''' so all the countries within the continent will be selected. You will also be able to search for a specific country using the '''Search Country''' field. If you call to a country not allowed in this section, the call will not be connected.&lt;br /&gt;
&lt;br /&gt;
Please make sure to only allow those destinations that you are planning to dial and avoid having unplanned destinations allowed.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:AllowCalls.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
 '''Note''': If you have international calls disabled, it will prevent calling to the international countries you have enabled in this list.&lt;br /&gt;
&lt;br /&gt;
== General ==&lt;br /&gt;
&lt;br /&gt;
These are the general settings used by the system when you make or receive calls.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:AcctSet Gen.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
'''e911 Default CallerID''': This enables you to use any CallerID you want to with your accounts but when you call E911 the number you specify here is used as your CallerID and the call is placed and completed successfully.&lt;br /&gt;
&lt;br /&gt;
'''Dialing Mode''': This setting allows you to set the way you're going to dial other numbers.&lt;br /&gt;
&lt;br /&gt;
*North America (Recommended): You can dial to countries part of the North American Numbering Plan Administration by dialing 10 or 11 digits. (with or without the 1 prefix).  You'll need to use the  00 or 011 prefix to call international numbers.&lt;br /&gt;
&lt;br /&gt;
*E164: You dial numbers by using Country Code first. You don't need to send the 00 or 011 prefix when using this mode, but it's still supported. &lt;br /&gt;
&lt;br /&gt;
'''CallerID Number''':  You can set the [[Caller ID|callerID Number]] you want to pass if you are using an [[Devices|ATA]], IP Phone or [[Softphones|Softphone]]. It is important to pass a valid [[Caller ID|caller id]] to ensure proper termination. If you have a a device capable of passing its own [[Caller ID|CallerID number]] such as a soft switch or [[PBXs|PBX]], you can leave this blank to set the [[Caller ID|CallerID Number]] on your side.&lt;br /&gt;
&lt;br /&gt;
'''Voicemail Associated to the Main Account''': You can select the mailbox associated to the main account. &lt;br /&gt;
&lt;br /&gt;
*Accessing Voice Mail: When you dial *97 from your main account, it will not prompt for the [[voicemail]] ID. It will also not prompt for the password if you have selected &amp;quot;Skip Password&amp;quot; option in the [[voicemail]] configuration. &lt;br /&gt;
&lt;br /&gt;
*Message Waiting Indicator: When there are new messages, notice of new message(s) will be sent. This will lead to different results depending on your type of adapter, soft phone or IP phone. For example, when using a  Linksys [[Devices|ATA]] adapter, it's usually a stutter tone with periodic ring, IP phones usually use a stutter dial tone when you pick up the line and are equipped with a blinking light, soft phones usually show up a Voicemail Icon in the dial display.&lt;br /&gt;
&lt;br /&gt;
 '''Note:'''This setting only affects the device registered with the main account. &lt;br /&gt;
      If you're using a [[Sub Accounts|sub account]] you need to set the '''Internal Extension Voicemail''' for that subaccount.&lt;br /&gt;
&lt;br /&gt;
'''Music On Hold''': Most IP Phones and [[Softphones|Softphones]] inform the VoIP server they are connected to when the HOLD Button is pressed. If you would like to present music to the person you place on hold, you can select it here.&amp;lt;br&amp;gt;&lt;br /&gt;
The following options for music are available:&amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
*'''No Music, Intermittent Bleep:''' Silent, but with a subtle intermittent bleep sound to let your callers know that they are still on the line while they hold.&lt;br /&gt;
*'''Away in the Tropics:''' From Hawaii to the Caribbean, these tracks deliver sounds of ukulele, steel drums, and steel guitars. &lt;br /&gt;
*'''Coffee and Sunrise:''' Uplifting without being perky, and positive without being too smiley. &lt;br /&gt;
*'''Coffee Shop Acoustic:''' Soothing, acoustic guitar tracks makes for a relaxed feel and sets a great atmosphere. &lt;br /&gt;
*'''Easy Listening:''' Smooth, casual tunes. &lt;br /&gt;
*'''Guitar Alchemy:''' Clever harmonics and progressive chord sequences to create a joyful and warming musical experience. &lt;br /&gt;
*'''Happy Endings:''' Uplifting, commercial style. Guitar, drums, ukulele, harmonica and bells. &lt;br /&gt;
*'''Light and Casual:''' Soothing and peaceful songs with a light, positive feeling. &lt;br /&gt;
*'''Orchestral Moods:''' Emotional and dramatic tales spun by violins, pianos and full orchestras. &lt;br /&gt;
*'''Piano Mix:'''  Smooth Piano &lt;br /&gt;
*'''Rock Me Easy:''' Feel-good music to create a relaxing atmosphere. &lt;br /&gt;
*'''Spa Sounds:''' Soft, slow and serene instrumentals. &lt;br /&gt;
&lt;br /&gt;
You can test the categories by dialing the following codes: &amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
'''*** 89''' Away in the Tropics &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 90''' Coffee and Sunrise &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 91''' Coffee Shop Acoustic &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 92''' Easy Listening &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 93''' Guitar Alchemy &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 94''' Happy Endings &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 95''' Light and Casual &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 96''' Orchestral Moods &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 97''' Piano Mix &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 98''' Rock Me Easy &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 99''' Spa Sounds&amp;lt;br&amp;gt;&lt;br /&gt;
'''***100''' No Music, Intermittent Bleep&amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
== Security ==&lt;br /&gt;
&lt;br /&gt;
Here you can change the password to access your Customer Portal as well your SIP/IAX passwords for your main account. You can also change the way SIP/IAX passwords are displayed on various pages of the &amp;quot;Customer Portal&amp;quot;.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accset sec.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Display SIP and IAX password(s) in Customer Portal''': If you enable this, the SIP/IAX passwords are going to be displayed in the Customer Portal. It is recommended that you leave this disabled. Enable this if only need if you forgot a SIP/IAX password for your account or one of your subaccounts. &lt;br /&gt;
 Note: You must enter your current customer portal password in order to enable this option.&lt;br /&gt;
&lt;br /&gt;
'''Customer Portal Password''': Here you can change the password used to log in your VoIP.ms Customer Portal.&lt;br /&gt;
&lt;br /&gt;
'''Main SIP/IAX Password''': Here is where you can change the password used to register your Main Account to one of the VoIP.ms servers.&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' For default the Main SIP/IAX Password is the same you use to login in your Customer Portal.&lt;br /&gt;
&lt;br /&gt;
== Inbound Settings==&lt;br /&gt;
&lt;br /&gt;
In this tab, you can set the protocol used for inbound calls and the device you're using for your main account.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Inbound.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Protocol for inbound DIDs''': Here you can select either SIP or IAX2 protocol. This will depend on the device you want to use with your Main account. SIP is the recommended protocol.&lt;br /&gt;
&lt;br /&gt;
'''Device type''': Make sure to select the correct type of device you're going to use with the main account to properly receive incoming calls in your device.&lt;br /&gt;
&lt;br /&gt;
== Notifications ==&lt;br /&gt;
&lt;br /&gt;
The notifications tab holds a complete list of alerts based upon most of the features that you will be able to see in the customer portal, in regards of changes over your configuration system structure, e911 dialing and API configuration changes.&lt;br /&gt;
&lt;br /&gt;
You will be able to select rather to receive an email or not when these events occur within your configuration, added to the always useful Balance Threshold.&lt;br /&gt;
&lt;br /&gt;
[[File:Accsettings2.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Balance Threshold''': Here you can set an amount and when your balance goes below this amount you will receive an email to the address set in the option below. You can select a value between $1 up to $200, or if you like you can disable the notifications (this is not recommended).&lt;br /&gt;
&lt;br /&gt;
 '''Note''': Set this according to the monthly use you have in your account. &lt;br /&gt;
&lt;br /&gt;
'''Email''': Enter the email address in which you want to receive the notifications. Make sure to enter a proper email address. If you want to receive notifications in more than one email address, you can list the emails separated with commas (,). This will only work for the Balance Threshold&lt;br /&gt;
&lt;br /&gt;
 '''Note''': While the balance is below your threshold, you're going to keep receiving emails until you add new funds in your account or you change the balance threshold.&lt;br /&gt;
&lt;br /&gt;
== Default DID Routing ==&lt;br /&gt;
&lt;br /&gt;
These settings let you define the preconfigured Routing options that are going to be applied to each new DIDs that you order. However, you are still able to change the settings for the DID in the order page.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:DefaultDIDRouting.png|800px]]&lt;br /&gt;
&lt;br /&gt;
'''Select Plan''': Incoming Calls and Monthly Fee for the DID will be charged according to the Plan selected here. Plan prices change depending on the rate center. These prices will be shown when purchasing numbers.&lt;br /&gt;
&lt;br /&gt;
'''CallerID Name Lookup''': When activated, the system will perform a lookup in the LIBD/CNAM database to find the name matching the number of callers with a US/Canadian CallerID Number. The result of this query will be displayed with the following format &amp;quot;Caller ID name portion&amp;quot; &amp;lt;Caller ID number&amp;gt;.&lt;br /&gt;
&lt;br /&gt;
'''DID POP''': Here you can set the server in which your software/device is registered to or receiving the call from.&lt;br /&gt;
 '''Note''': Always make sure to register your software/device to the same server you set in this option to properly receive incoming calls.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:DefaultRouting.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Routing''': Here you can select the destination of the incoming calls to your DID number. You can routed directly to one account/sub account, an [[Digital Receptionist (IVR)]], [[Calling Queues]], [[Time Conditions]], [[Ring Groups]], etc.&lt;br /&gt;
&lt;br /&gt;
'''Failover Options''': This setting let you define where to redirect the call when the destination is '''Busy''', '''Unreachable''' or '''No Answer'''. These options can be displayed by clicking the '''Show Failover Options''' button&lt;br /&gt;
&lt;br /&gt;
 Note: You can change these settings later or enable additional settings in your Customer Portal&amp;gt;&amp;gt;DID Numbers&amp;gt;&amp;gt;Manage DID(s)&lt;br /&gt;
&lt;br /&gt;
== Newsletter ==&lt;br /&gt;
&lt;br /&gt;
'''Newsletter Subscription''': You can set if you want to receive news from voip.ms&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accsettings3.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
== Advanced ==&lt;br /&gt;
&lt;br /&gt;
These options allow you to set the allowed codecs, DTMF modes and NAT options. These settings are recommended for advanced users only.&lt;br /&gt;
&lt;br /&gt;
[[File:Accset adv.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''NAT (Network Address Translation)''': This setting should be set to Yes if you're behind a NAT, if not set to No. If you're unsure what setting means, is highly recommended that you leave it to Yes.&lt;br /&gt;
&lt;br /&gt;
'''DTMF Mode''': This allows you to select the DTMF mode that is going to be used for your account. If you set this to AUTO the RFC2833 (AVT) is going to be used and automatically switch to INBAND if the other end doesn't support RFC2833.&lt;br /&gt;
 Note: Its recommended that you select the same DTMF mode in your device. &lt;br /&gt;
&lt;br /&gt;
'''Allowed Codecs''': This setting allows you to select the codecs that you're going to use, you can choose to allow one or more codecs at the time.&lt;br /&gt;
 Note: It's recommended that you select Allow All and only change it if you have an specific reason to do so.&lt;br /&gt;
&lt;br /&gt;
[[category:guides]]&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/Account_Settings</id>
		<title>Account Settings</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/Account_Settings"/>
				<updated>2017-02-14T21:35:15Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;The Account Settings page is one of the most important pages in the Customer Portal, it will help you out as its name points out, to set all the proper ways to use your account, and these settings will affect the global settings on your account, from Device type to routing and even security of your account. &lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
__TOC__&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
== Account Routing ==&lt;br /&gt;
&lt;br /&gt;
These settings define the routes the system will use when you place calls to either US, Canada, Toll Free or International Numbers.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accsettings1.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Canada Routing''': You can choose between the [[Value vs Premium|'''Value''' or '''Premium''']] Route. If you are looking for the best wholesale price without any volume commitment you can use the '''Value''' route, however the CallerID number is not always guaranteed to work in this route.&lt;br /&gt;
The '''Premium''' route has a better level of quality, at a price that is a little higher than the value option. The CallerID Number is guaranteed to work for this route. Please note that the '''Canada Route''' can also be set independently per sub account, under the [[Sub Accounts]] settings. &lt;br /&gt;
&lt;br /&gt;
  Please note the US route is only available on the Premium route with a rate of $0.009 per minute.&lt;br /&gt;
&lt;br /&gt;
'''International Routing''': You can choose also the [[Value vs Premium|'''Value''' or '''Premium''']] route. Both routes are reliable, however you're going to find better quality if you're using the '''Premium''' route. The CallerID Number is not guaranteed to work for international calls even if you use the '''Premium''' route.&lt;br /&gt;
&lt;br /&gt;
'''Toll Free Routing''': This define the route the system will use when you place a Toll-Free call to Canada or USA. You can choose between the '''Value''' route (free to use) or the '''Premium''' (at 1.06 cent per minute). As the other settings, the '''Premium''' route give you better quality and the CallerID Number is guaranteed to work.&lt;br /&gt;
&lt;br /&gt;
'''Toll Free Carrier''': This setting defines the type of carrier that will be used when you dial a Toll-Free number. You can choose between '''Server Default''', '''American Carrier''' or '''Canadian Carrier'''. &lt;br /&gt;
&lt;br /&gt;
 '''Server Default:''' The Carrier for the termination of your toll-free call will be chosen depending on the geographical location of the server &lt;br /&gt;
   you are registering or connecting to. American Servers will dial through an American Carrier and Canadian servers will use a Canadian Toll-Free carrier.&lt;br /&gt;
 '''American Carrier:''' Your Toll-Free calls will always go through an American Carrier, even if you use a Canadian server. &lt;br /&gt;
 '''Canadian Carrier:''' Your Toll-Free calls will always go through a Canadian Carrier, even if you use an American server.&lt;br /&gt;
&lt;br /&gt;
For example, this feature can be useful for west based Canadian customers who wish to use the Seattle server but have their toll-free termination calls go through a Canadian carrier.&lt;br /&gt;
&lt;br /&gt;
 '''Note''': Please refer to [[Service Cost]] for information about the rates of the outgoing calls.&lt;br /&gt;
&lt;br /&gt;
== Account Restrictions ==&lt;br /&gt;
&lt;br /&gt;
These settings define the restrictions the system will use when you place calls to either USA, Canada or International Numbers.&lt;br /&gt;
&lt;br /&gt;
[[File:Account Restrictions.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Allow 411 dialing''': When enabled, you can call 411 directory assistance at the cost of $0.99 per call. &lt;br /&gt;
&lt;br /&gt;
 '''Note''': XXX-555-1212 is disabled within VoIP.ms. The only way to reach directory assistance is by dialing 411.&lt;br /&gt;
&lt;br /&gt;
'''Allow International Calls''': When Disabled, calls placed to destinations outside USA/Canada will be automatically rejected. If you are using [[Sub Accounts]] you need to setup this directly in the sub account.&lt;br /&gt;
&lt;br /&gt;
 '''Note''': This option is set to &amp;quot;No&amp;quot; by default for New accounts.&lt;br /&gt;
&lt;br /&gt;
'''Max. Call Time for US48/Canadian Calls''': With these setting you can choose how long the call will last when you call to US or Canada Numbers. If a call to a US or Canada Number exceeds the maximum time you have set in this option, it will be automatically hung up.&lt;br /&gt;
&lt;br /&gt;
'''Max. Call Time for International Calls''': Works exactly as the '''Max. Call Time for US48/Canadian Calls''' setting. If a call to an International Number exceeds the maximum time set, it will be automatically hung up.&lt;br /&gt;
&lt;br /&gt;
'''International Amount Restriction''': With this option you can set the maximum amount per minute for calls to international destinations. If the Per minute rate for an International destination exceeds the value you have set in this option, the call will not be connected.&lt;br /&gt;
&lt;br /&gt;
For example, let say you can set this option to $0.150 and you want to call a UK mobile number with area code 4470, the call will not go through because the per minute rate for this area code is $0.3048.&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' This settings depends on the International Routing you have choose.&lt;br /&gt;
&lt;br /&gt;
'''Allow Calls to Countries''': Here you can select Regions or Specific Countries to allow outgoing calls from your account. &lt;br /&gt;
&lt;br /&gt;
In order to allow calls to a specific country you can click the name of the region to expand the countries within it and select the desired countries only. You can select '''Select All''' so all the countries within the continent will be selected. You will also be able to search for a specific country using the '''Search Country''' field. If you call to a country not allowed in this section, the call will not be connected.&lt;br /&gt;
&lt;br /&gt;
Please make sure to only allow those destinations that you are planning to dial and avoid having unplanned destinations allowed.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:AllowCalls.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
 '''Note''': If you have international calls disabled, it will prevent calling to the international countries you have enabled in this list.&lt;br /&gt;
&lt;br /&gt;
== General ==&lt;br /&gt;
&lt;br /&gt;
These are the general settings used by the system when you make or receive calls.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:AcctSet Gen.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
'''e911 Default CallerID''': This enables you to use any CallerID you want to with your accounts but when you call E911 the number you specify here is used as your CallerID and the call is placed and completed successfully.&lt;br /&gt;
&lt;br /&gt;
'''Dialing Mode''': This setting allows you to set the way you're going to dial other numbers.&lt;br /&gt;
&lt;br /&gt;
*North America (Recommended): You can dial to countries part of the North American Numbering Plan Administration by dialing 10 or 11 digits. (with or without the 1 prefix).  You'll need to use the  00 or 011 prefix to call international numbers.&lt;br /&gt;
&lt;br /&gt;
*E164: You dial numbers by using Country Code first. You don't need to send the 00 or 011 prefix when using this mode, but it's still supported. &lt;br /&gt;
&lt;br /&gt;
'''CallerID Number''':  You can set the [[Caller ID|callerID Number]] you want to pass if you are using an [[Devices|ATA]], IP Phone or [[Softphones|Softphone]]. It is important to pass a valid [[Caller ID|caller id]] to ensure proper termination. If you have a a device capable of passing its own [[Caller ID|CallerID number]] such as a soft switch or [[PBXs|PBX]], you can leave this blank to set the [[Caller ID|CallerID Number]] on your side.&lt;br /&gt;
&lt;br /&gt;
'''Voicemail Associated to the Main Account''': You can select the mailbox associated to the main account. &lt;br /&gt;
&lt;br /&gt;
*Accessing Voice Mail: When you dial *97 from your main account, it will not prompt for the [[voicemail]] ID. It will also not prompt for the password if you have selected &amp;quot;Skip Password&amp;quot; option in the [[voicemail]] configuration. &lt;br /&gt;
&lt;br /&gt;
*Message Waiting Indicator: When there are new messages, notice of new message(s) will be sent. This will lead to different results depending on your type of adapter, soft phone or IP phone. For example, when using a  Linksys [[Devices|ATA]] adapter, it's usually a stutter tone with periodic ring, IP phones usually use a stutter dial tone when you pick up the line and are equipped with a blinking light, soft phones usually show up a Voicemail Icon in the dial display.&lt;br /&gt;
&lt;br /&gt;
 '''Note:'''This setting only affects the device registered with the main account. &lt;br /&gt;
      If you're using a [[Sub Accounts|sub account]] you need to set the '''Internal Extension Voicemail''' for that subaccount.&lt;br /&gt;
&lt;br /&gt;
'''Music On Hold''': Most IP Phones and [[Softphones|Softphones]] inform the VoIP server they are connected to when the HOLD Button is pressed. If you would like to present music to the person you place on hold, you can select it here.&amp;lt;br&amp;gt;&lt;br /&gt;
The following options for music are available:&amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
*'''No Music, Intermittent Bleep:''' Silent, but with a subtle intermittent bleep sound to let your callers know that they are still on the line while they hold.&lt;br /&gt;
*'''Away in the Tropics:''' From Hawaii to the Caribbean, these tracks deliver sounds of ukulele, steel drums, and steel guitars. &lt;br /&gt;
*'''Coffee and Sunrise:''' Uplifting without being perky, and positive without being too smiley. &lt;br /&gt;
*'''Coffee Shop Acoustic:''' Soothing, acoustic guitar tracks makes for a relaxed feel and sets a great atmosphere. &lt;br /&gt;
*'''Easy Listening:''' Smooth, casual tunes. &lt;br /&gt;
*'''Guitar Alchemy:''' Clever harmonics and progressive chord sequences to create a joyful and warming musical experience. &lt;br /&gt;
*'''Happy Endings:''' Uplifting, commercial style. Guitar, drums, ukulele, harmonica and bells. &lt;br /&gt;
*'''Light and Casual:''' Soothing and peaceful songs with a light, positive feeling. &lt;br /&gt;
*'''Orchestral Moods:''' Emotional and dramatic tales spun by violins, pianos and full orchestras. &lt;br /&gt;
*'''Piano Mix:'''  Smooth Piano &lt;br /&gt;
*'''Rock Me Easy:''' Feel-good music to create a relaxing atmosphere. &lt;br /&gt;
*'''Spa Sounds:''' Soft, slow and serene instrumentals. &lt;br /&gt;
&lt;br /&gt;
You can test the categories by dialing the following codes: &amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
'''*** 89''' Away in the Tropics &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 90''' Coffee and Sunrise &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 91''' Coffee Shop Acoustic &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 92''' Easy Listening &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 93''' Guitar Alchemy &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 94''' Happy Endings &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 95''' Light and Casual &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 96''' Orchestral Moods &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 97''' Piano Mix &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 98''' Rock Me Easy &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 99''' Spa Sounds&amp;lt;br&amp;gt;&lt;br /&gt;
'''***100''' No Music, Intermittent Bleep&amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
== Security ==&lt;br /&gt;
&lt;br /&gt;
Here you can change the password to access your Customer Portal as well your SIP/IAX passwords for your main account. You can also change the way SIP/IAX passwords are displayed on various pages of the &amp;quot;Customer Portal&amp;quot;.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accset sec.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Display SIP and IAX password(s) in Customer Portal''': If you enable this, the SIP/IAX passwords are going to be displayed in the Customer Portal. It is recommended that you leave this disabled. Enable this if only need if you forgot a SIP/IAX password for your account or one of your subaccounts. &lt;br /&gt;
 Note: You must enter your current customer portal password in order to enable this option.&lt;br /&gt;
&lt;br /&gt;
'''Customer Portal Password''': Here you can change the password used to log in your VoIP.ms Customer Portal.&lt;br /&gt;
&lt;br /&gt;
'''Main SIP/IAX Password''': Here is where you can change the password used to register your Main Account to one of the VoIP.ms servers.&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' For default the Main SIP/IAX Password is the same you use to login in your Customer Portal.&lt;br /&gt;
&lt;br /&gt;
== Inbound Settings==&lt;br /&gt;
&lt;br /&gt;
In this tab, you can set the protocol used for inbound calls and the device you're using for your main account.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Inbound.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Protocol for inbound DIDs''': Here you can select either SIP or IAX2 protocol. This will depend on the device you want to use with your Main account. SIP is the recommended protocol.&lt;br /&gt;
&lt;br /&gt;
'''Device type''': Make sure to select the correct type of device you're going to use with the main account to properly receive incoming calls in your device.&lt;br /&gt;
&lt;br /&gt;
== Notifications ==&lt;br /&gt;
&lt;br /&gt;
The notifications tab holds a complete list of alerts based upon most of the features that you will be able to see in the customer portal, in regards of changes over your configuration system structure, e911 dialing and API configuration changes. You will be able to select rather to receive an email or not when these events occur within your configuration, added to the always useful Balance Threshold.&lt;br /&gt;
&lt;br /&gt;
[[File:Accsettings2.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Balance Threshold''': Here you can set an amount and when your balance goes below this amount you will receive an email to the address set in the option below. You can select a value between $1 up to $200, or if you like you can disable the notifications (this is not recommended).&lt;br /&gt;
&lt;br /&gt;
 '''Note''': Set this according to the monthly use you have in your account. &lt;br /&gt;
&lt;br /&gt;
'''Email''': Enter the email address in which you want to receive the notifications. Make sure to enter a proper email address. If you want to receive notifications in more than one email address, you can list the emails separated with commas (,). This will only work for the Balance Threshold&lt;br /&gt;
&lt;br /&gt;
 '''Note''': While the balance is below your threshold, you're going to keep receiving emails until you add new funds in your account or you change the balance threshold.&lt;br /&gt;
&lt;br /&gt;
== Default DID Routing ==&lt;br /&gt;
&lt;br /&gt;
These settings let you define the preconfigured Routing options that are going to be applied to each new DIDs that you order. However, you are still able to change the settings for the DID in the order page.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:DefaultDIDRouting.png|800px]]&lt;br /&gt;
&lt;br /&gt;
'''Select Plan''': Incoming Calls and Monthly Fee for the DID will be charged according to the Plan selected here. Plan prices change depending on the rate center. These prices will be shown when purchasing numbers.&lt;br /&gt;
&lt;br /&gt;
'''CallerID Name Lookup''': When activated, the system will perform a lookup in the LIBD/CNAM database to find the name matching the number of callers with a US/Canadian CallerID Number. The result of this query will be displayed with the following format &amp;quot;Caller ID name portion&amp;quot; &amp;lt;Caller ID number&amp;gt;.&lt;br /&gt;
&lt;br /&gt;
'''DID POP''': Here you can set the server in which your software/device is registered to or receiving the call from.&lt;br /&gt;
 '''Note''': Always make sure to register your software/device to the same server you set in this option to properly receive incoming calls.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:DefaultRouting.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Routing''': Here you can select the destination of the incoming calls to your DID number. You can routed directly to one account/sub account, an [[Digital Receptionist (IVR)]], [[Calling Queues]], [[Time Conditions]], [[Ring Groups]], etc.&lt;br /&gt;
&lt;br /&gt;
'''Failover Options''': This setting let you define where to redirect the call when the destination is '''Busy''', '''Unreachable''' or '''No Answer'''. These options can be displayed by clicking the '''Show Failover Options''' button&lt;br /&gt;
&lt;br /&gt;
 Note: You can change these settings later or enable additional settings in your Customer Portal&amp;gt;&amp;gt;DID Numbers&amp;gt;&amp;gt;Manage DID(s)&lt;br /&gt;
&lt;br /&gt;
== Newsletter ==&lt;br /&gt;
&lt;br /&gt;
'''Newsletter Subscription''': You can set if you want to receive news from voip.ms&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accsettings3.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
== Advanced ==&lt;br /&gt;
&lt;br /&gt;
These options allow you to set the allowed codecs, DTMF modes and NAT options. These settings are recommended for advanced users only.&lt;br /&gt;
&lt;br /&gt;
[[File:Accset adv.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''NAT (Network Address Translation)''': This setting should be set to Yes if you're behind a NAT, if not set to No. If you're unsure what setting means, is highly recommended that you leave it to Yes.&lt;br /&gt;
&lt;br /&gt;
'''DTMF Mode''': This allows you to select the DTMF mode that is going to be used for your account. If you set this to AUTO the RFC2833 (AVT) is going to be used and automatically switch to INBAND if the other end doesn't support RFC2833.&lt;br /&gt;
 Note: Its recommended that you select the same DTMF mode in your device. &lt;br /&gt;
&lt;br /&gt;
'''Allowed Codecs''': This setting allows you to select the codecs that you're going to use, you can choose to allow one or more codecs at the time.&lt;br /&gt;
 Note: It's recommended that you select Allow All and only change it if you have an specific reason to do so.&lt;br /&gt;
&lt;br /&gt;
[[category:guides]]&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/Account_Settings</id>
		<title>Account Settings</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/Account_Settings"/>
				<updated>2017-02-14T21:24:14Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;The Account Settings page is one of the most important pages in the Customer Portal, it will help you out as its name points out, to set all the proper ways to use your account, and these settings will affect the global settings on your account, from Device type to routing and even security of your account. &lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
__TOC__&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
== Account Routing ==&lt;br /&gt;
&lt;br /&gt;
These settings define the routes the system will use when you place calls to either US, Canada, Toll Free or International Numbers.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accsettings1.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Canada Routing''': You can choose between the [[Value vs Premium|'''Value''' or '''Premium''']] Route. If you are looking for the best wholesale price without any volume commitment you can use the '''Value''' route, however the CallerID number is not always guaranteed to work in this route.&lt;br /&gt;
The '''Premium''' route has a better level of quality, at a price that is a little higher than the value option. The CallerID Number is guaranteed to work for this route. Please note that the '''Canada Route''' can also be set independently per sub account, under the [[Sub Accounts]] settings. &lt;br /&gt;
&lt;br /&gt;
  Please note the US route is only available on the Premium route with a rate of $0.009 per minute.&lt;br /&gt;
&lt;br /&gt;
'''International Routing''': You can choose also the [[Value vs Premium|'''Value''' or '''Premium''']] route. Both routes are reliable, however you're going to find better quality if you're using the '''Premium''' route. The CallerID Number is not guaranteed to work for international calls even if you use the '''Premium''' route.&lt;br /&gt;
&lt;br /&gt;
'''Toll Free Routing''': This define the route the system will use when you place a Toll-Free call to Canada or USA. You can choose between the '''Value''' route (free to use) or the '''Premium''' (at 1.06 cent per minute). As the other settings, the '''Premium''' route give you better quality and the CallerID Number is guaranteed to work.&lt;br /&gt;
&lt;br /&gt;
'''Toll Free Carrier''': This setting defines the type of carrier that will be used when you dial a Toll-Free number. You can choose between '''Server Default''', '''American Carrier''' or '''Canadian Carrier'''. &lt;br /&gt;
&lt;br /&gt;
 '''Server Default:''' The Carrier for the termination of your toll-free call will be chosen depending on the geographical location of the server &lt;br /&gt;
   you are registering or connecting to. American Servers will dial through an American Carrier and Canadian servers will use a Canadian Toll-Free carrier.&lt;br /&gt;
 '''American Carrier:''' Your Toll-Free calls will always go through an American Carrier, even if you use a Canadian server. &lt;br /&gt;
 '''Canadian Carrier:''' Your Toll-Free calls will always go through a Canadian Carrier, even if you use an American server.&lt;br /&gt;
&lt;br /&gt;
For example, this feature can be useful for west based Canadian customers who wish to use the Seattle server but have their toll-free termination calls go through a Canadian carrier.&lt;br /&gt;
&lt;br /&gt;
 '''Note''': Please refer to [[Service Cost]] for information about the rates of the outgoing calls.&lt;br /&gt;
&lt;br /&gt;
== Account Restrictions ==&lt;br /&gt;
&lt;br /&gt;
These settings define the restrictions the system will use when you place calls to either USA, Canada or International Numbers.&lt;br /&gt;
&lt;br /&gt;
[[File:Account Restrictions.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Allow 411 dialing''': When enabled, you can call 411 directory assistance at the cost of $0.99 per call. &lt;br /&gt;
&lt;br /&gt;
 '''Note''': XXX-555-1212 is disabled within VoIP.ms. The only way to reach directory assistance is by dialing 411.&lt;br /&gt;
&lt;br /&gt;
'''Allow International Calls''': When Disabled, calls placed to destinations outside USA/Canada will be automatically rejected. If you are using [[Sub Accounts]] you need to setup this directly in the sub account.&lt;br /&gt;
&lt;br /&gt;
 '''Note''': This option is set to &amp;quot;No&amp;quot; by default for New accounts.&lt;br /&gt;
&lt;br /&gt;
'''Max. Call Time for US48/Canadian Calls''': With these setting you can choose how long the call will last when you call to US or Canada Numbers. If a call to a US or Canada Number exceeds the maximum time you have set in this option, it will be automatically hung up.&lt;br /&gt;
&lt;br /&gt;
'''Max. Call Time for International Calls''': Works exactly as the '''Max. Call Time for US48/Canadian Calls''' setting. If a call to an International Number exceeds the maximum time set, it will be automatically hung up.&lt;br /&gt;
&lt;br /&gt;
'''International Amount Restriction''': With this option you can set the maximum amount per minute for calls to international destinations. If the Per minute rate for an International destination exceeds the value you have set in this option, the call will not be connected.&lt;br /&gt;
&lt;br /&gt;
For example, let say you can set this option to $0.150 and you want to call a UK mobile number with area code 4470, the call will not go through because the per minute rate for this area code is $0.3048.&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' This settings depends on the International Routing you have choose.&lt;br /&gt;
&lt;br /&gt;
'''Allow Calls to Countries''': Here you can select Regions or Specific Countries to allow outgoing calls from your account. &lt;br /&gt;
&lt;br /&gt;
In order to allow calls to a specific country you can click the name of the region to expand the countries within it and select the desired countries only. You can select '''Select All''' so all the countries within the continent will be selected. You will also be able to search for a specific country using the '''Search Country''' field. If you call to a country not allowed in this section, the call will not be connected.&lt;br /&gt;
&lt;br /&gt;
Please make sure to only allow those destinations that you are planning to dial and avoid having unplanned destinations allowed.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:AllowCalls.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
 '''Note''': If you have international calls disabled, it will prevent calling to the international countries you have enabled in this list.&lt;br /&gt;
&lt;br /&gt;
== General ==&lt;br /&gt;
&lt;br /&gt;
These are the general settings used by the system when you make or receive calls.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:AcctSet Gen.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
'''e911 Default CallerID''': This enables you to use any CallerID you want to with your accounts but when you call E911 the number you specify here is used as your CallerID and the call is placed and completed successfully.&lt;br /&gt;
&lt;br /&gt;
'''Dialing Mode''': This setting allows you to set the way you're going to dial other numbers.&lt;br /&gt;
&lt;br /&gt;
*North America (Recommended): You can dial to countries part of the North American Numbering Plan Administration by dialing 10 or 11 digits. (with or without the 1 prefix).  You'll need to use the  00 or 011 prefix to call international numbers.&lt;br /&gt;
&lt;br /&gt;
*E164: You dial numbers by using Country Code first. You don't need to send the 00 or 011 prefix when using this mode, but it's still supported. &lt;br /&gt;
&lt;br /&gt;
'''CallerID Number''':  You can set the [[Caller ID|callerID Number]] you want to pass if you are using an [[Devices|ATA]], IP Phone or [[Softphones|Softphone]]. It is important to pass a valid [[Caller ID|caller id]] to ensure proper termination. If you have a a device capable of passing its own [[Caller ID|CallerID number]] such as a soft switch or [[PBXs|PBX]], you can leave this blank to set the [[Caller ID|CallerID Number]] on your side.&lt;br /&gt;
&lt;br /&gt;
'''Voicemail Associated to the Main Account''': You can select the mailbox associated to the main account. &lt;br /&gt;
&lt;br /&gt;
*Accessing Voice Mail: When you dial *97 from your main account, it will not prompt for the [[voicemail]] ID. It will also not prompt for the password if you have selected &amp;quot;Skip Password&amp;quot; option in the [[voicemail]] configuration. &lt;br /&gt;
&lt;br /&gt;
*Message Waiting Indicator: When there are new messages, notice of new message(s) will be sent. This will lead to different results depending on your type of adapter, soft phone or IP phone. For example, when using a  Linksys [[Devices|ATA]] adapter, it's usually a stutter tone with periodic ring, IP phones usually use a stutter dial tone when you pick up the line and are equipped with a blinking light, soft phones usually show up a Voicemail Icon in the dial display.&lt;br /&gt;
&lt;br /&gt;
 '''Note:'''This setting only affects the device registered with the main account. &lt;br /&gt;
      If you're using a [[Sub Accounts|sub account]] you need to set the '''Internal Extension Voicemail''' for that subaccount.&lt;br /&gt;
&lt;br /&gt;
'''Music On Hold''': Most IP Phones and [[Softphones|Softphones]] inform the VoIP server they are connected to when the HOLD Button is pressed. If you would like to present music to the person you place on hold, you can select it here.&amp;lt;br&amp;gt;&lt;br /&gt;
The following options for music are available:&amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
*'''No Music, Intermittent Bleep:''' Silent, but with a subtle intermittent bleep sound to let your callers know that they are still on the line while they hold.&lt;br /&gt;
*'''Away in the Tropics:''' From Hawaii to the Caribbean, these tracks deliver sounds of ukulele, steel drums, and steel guitars. &lt;br /&gt;
*'''Coffee and Sunrise:''' Uplifting without being perky, and positive without being too smiley. &lt;br /&gt;
*'''Coffee Shop Acoustic:''' Soothing, acoustic guitar tracks makes for a relaxed feel and sets a great atmosphere. &lt;br /&gt;
*'''Easy Listening:''' Smooth, casual tunes. &lt;br /&gt;
*'''Guitar Alchemy:''' Clever harmonics and progressive chord sequences to create a joyful and warming musical experience. &lt;br /&gt;
*'''Happy Endings:''' Uplifting, commercial style. Guitar, drums, ukulele, harmonica and bells. &lt;br /&gt;
*'''Light and Casual:''' Soothing and peaceful songs with a light, positive feeling. &lt;br /&gt;
*'''Orchestral Moods:''' Emotional and dramatic tales spun by violins, pianos and full orchestras. &lt;br /&gt;
*'''Piano Mix:'''  Smooth Piano &lt;br /&gt;
*'''Rock Me Easy:''' Feel-good music to create a relaxing atmosphere. &lt;br /&gt;
*'''Spa Sounds:''' Soft, slow and serene instrumentals. &lt;br /&gt;
&lt;br /&gt;
You can test the categories by dialing the following codes: &amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
'''*** 89''' Away in the Tropics &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 90''' Coffee and Sunrise &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 91''' Coffee Shop Acoustic &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 92''' Easy Listening &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 93''' Guitar Alchemy &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 94''' Happy Endings &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 95''' Light and Casual &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 96''' Orchestral Moods &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 97''' Piano Mix &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 98''' Rock Me Easy &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 99''' Spa Sounds&amp;lt;br&amp;gt;&lt;br /&gt;
'''***100''' No Music, Intermittent Bleep&amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
== Security ==&lt;br /&gt;
&lt;br /&gt;
Here you can change the password to access your Customer Portal as well your SIP/IAX passwords for your main account. You can also change the way SIP/IAX passwords are displayed on various pages of the &amp;quot;Customer Portal&amp;quot;.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accset sec.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Display SIP and IAX password(s) in Customer Portal''': If you enable this, the SIP/IAX passwords are going to be displayed in the Customer Portal. It is recommended that you leave this disabled. Enable this if only need if you forgot a SIP/IAX password for your account or one of your subaccounts. &lt;br /&gt;
 Note: You must enter your current customer portal password in order to enable this option.&lt;br /&gt;
&lt;br /&gt;
'''Customer Portal Password''': Here you can change the password used to log in your VoIP.ms Customer Portal.&lt;br /&gt;
&lt;br /&gt;
'''Main SIP/IAX Password''': Here is where you can change the password used to register your Main Account to one of the VoIP.ms servers.&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' For default the Main SIP/IAX Password is the same you use to login in your Customer Portal.&lt;br /&gt;
&lt;br /&gt;
== Inbound Settings==&lt;br /&gt;
&lt;br /&gt;
In this tab, you can set the protocol used for inbound calls and the device you're using for your main account.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Inbound.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Protocol for inbound DIDs''': Here you can select either SIP or IAX2 protocol. This will depend on the device you want to use with your Main account. SIP is the recommended protocol.&lt;br /&gt;
&lt;br /&gt;
'''Device type''': Make sure to select the correct type of device you're going to use with the main account to properly receive incoming calls in your device.&lt;br /&gt;
&lt;br /&gt;
== Notifications ==&lt;br /&gt;
&lt;br /&gt;
In this tab you can set whether to allow or not notifications in your mail when the balance in your account reach certain amount.&lt;br /&gt;
&lt;br /&gt;
[[File:Accsettings2.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Balance Threshold''': Here you can set an amount and when your balance goes below this amount you will receive an email to the address set in the option below. You can select a value between $1 up to $200, or if you like you can disable the notifications (this is not recommended).&lt;br /&gt;
&lt;br /&gt;
 '''Note''': Set this according to the monthly use you have in your account. &lt;br /&gt;
&lt;br /&gt;
'''Email''': Enter the email address in which you want to receive the notifications. Make sure to enter a proper email address. If you want to receive notifications in more than one email address, you can list the emails separated with commas (,). This will only work for the Balance Threshold&lt;br /&gt;
&lt;br /&gt;
 '''Note''': While the balance is below your threshold, you're going to keep receiving emails until you add new funds in your account or you change the balance threshold.&lt;br /&gt;
&lt;br /&gt;
== Default DID Routing ==&lt;br /&gt;
&lt;br /&gt;
These settings let you define the preconfigured Routing options that are going to be applied to each new DIDs that you order. However, you are still able to change the settings for the DID in the order page.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:DefaultDIDRouting.png|800px]]&lt;br /&gt;
&lt;br /&gt;
'''Select Plan''': Incoming Calls and Monthly Fee for the DID will be charged according to the Plan selected here. Plan prices change depending on the rate center. These prices will be shown when purchasing numbers.&lt;br /&gt;
&lt;br /&gt;
'''CallerID Name Lookup''': When activated, the system will perform a lookup in the LIBD/CNAM database to find the name matching the number of callers with a US/Canadian CallerID Number. The result of this query will be displayed with the following format &amp;quot;Caller ID name portion&amp;quot; &amp;lt;Caller ID number&amp;gt;.&lt;br /&gt;
&lt;br /&gt;
'''DID POP''': Here you can set the server in which your software/device is registered to or receiving the call from.&lt;br /&gt;
 '''Note''': Always make sure to register your software/device to the same server you set in this option to properly receive incoming calls.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:DefaultRouting.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Routing''': Here you can select the destination of the incoming calls to your DID number. You can routed directly to one account/sub account, an [[Digital Receptionist (IVR)]], [[Calling Queues]], [[Time Conditions]], [[Ring Groups]], etc.&lt;br /&gt;
&lt;br /&gt;
'''Failover Options''': This setting let you define where to redirect the call when the destination is '''Busy''', '''Unreachable''' or '''No Answer'''. These options can be displayed by clicking the '''Show Failover Options''' button&lt;br /&gt;
&lt;br /&gt;
 Note: You can change these settings later or enable additional settings in your Customer Portal&amp;gt;&amp;gt;DID Numbers&amp;gt;&amp;gt;Manage DID(s)&lt;br /&gt;
&lt;br /&gt;
== Newsletter ==&lt;br /&gt;
&lt;br /&gt;
'''Newsletter Subscription''': You can set if you want to receive news from voip.ms&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accsettings3.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
== Advanced ==&lt;br /&gt;
&lt;br /&gt;
These options allow you to set the allowed codecs, DTMF modes and NAT options. These settings are recommended for advanced users only.&lt;br /&gt;
&lt;br /&gt;
[[File:Accset adv.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''NAT (Network Address Translation)''': This setting should be set to Yes if you're behind a NAT, if not set to No. If you're unsure what setting means, is highly recommended that you leave it to Yes.&lt;br /&gt;
&lt;br /&gt;
'''DTMF Mode''': This allows you to select the DTMF mode that is going to be used for your account. If you set this to AUTO the RFC2833 (AVT) is going to be used and automatically switch to INBAND if the other end doesn't support RFC2833.&lt;br /&gt;
 Note: Its recommended that you select the same DTMF mode in your device. &lt;br /&gt;
&lt;br /&gt;
'''Allowed Codecs''': This setting allows you to select the codecs that you're going to use, you can choose to allow one or more codecs at the time.&lt;br /&gt;
 Note: It's recommended that you select Allow All and only change it if you have an specific reason to do so.&lt;br /&gt;
&lt;br /&gt;
[[category:guides]]&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/File:Accsettings3.png</id>
		<title>File:Accsettings3.png</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/File:Accsettings3.png"/>
				<updated>2017-02-14T21:23:44Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/Account_Settings</id>
		<title>Account Settings</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/Account_Settings"/>
				<updated>2017-02-14T21:22:14Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;The Account Settings page is one of the most important pages in the Customer Portal, it will help you out as its name points out, to set all the proper ways to use your account, and these settings will affect the global settings on your account, from Device type to routing and even security of your account. &lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
__TOC__&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
== Account Routing ==&lt;br /&gt;
&lt;br /&gt;
These settings define the routes the system will use when you place calls to either US, Canada, Toll Free or International Numbers.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accsettings1.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Canada Routing''': You can choose between the [[Value vs Premium|'''Value''' or '''Premium''']] Route. If you are looking for the best wholesale price without any volume commitment you can use the '''Value''' route, however the CallerID number is not always guaranteed to work in this route.&lt;br /&gt;
The '''Premium''' route has a better level of quality, at a price that is a little higher than the value option. The CallerID Number is guaranteed to work for this route. Please note that the '''Canada Route''' can also be set independently per sub account, under the [[Sub Accounts]] settings. &lt;br /&gt;
&lt;br /&gt;
  Please note the US route is only available on the Premium route with a rate of $0.009 per minute.&lt;br /&gt;
&lt;br /&gt;
'''International Routing''': You can choose also the [[Value vs Premium|'''Value''' or '''Premium''']] route. Both routes are reliable, however you're going to find better quality if you're using the '''Premium''' route. The CallerID Number is not guaranteed to work for international calls even if you use the '''Premium''' route.&lt;br /&gt;
&lt;br /&gt;
'''Toll Free Routing''': This define the route the system will use when you place a Toll-Free call to Canada or USA. You can choose between the '''Value''' route (free to use) or the '''Premium''' (at 1.06 cent per minute). As the other settings, the '''Premium''' route give you better quality and the CallerID Number is guaranteed to work.&lt;br /&gt;
&lt;br /&gt;
'''Toll Free Carrier''': This setting defines the type of carrier that will be used when you dial a Toll-Free number. You can choose between '''Server Default''', '''American Carrier''' or '''Canadian Carrier'''. &lt;br /&gt;
&lt;br /&gt;
 '''Server Default:''' The Carrier for the termination of your toll-free call will be chosen depending on the geographical location of the server &lt;br /&gt;
   you are registering or connecting to. American Servers will dial through an American Carrier and Canadian servers will use a Canadian Toll-Free carrier.&lt;br /&gt;
 '''American Carrier:''' Your Toll-Free calls will always go through an American Carrier, even if you use a Canadian server. &lt;br /&gt;
 '''Canadian Carrier:''' Your Toll-Free calls will always go through a Canadian Carrier, even if you use an American server.&lt;br /&gt;
&lt;br /&gt;
For example, this feature can be useful for west based Canadian customers who wish to use the Seattle server but have their toll-free termination calls go through a Canadian carrier.&lt;br /&gt;
&lt;br /&gt;
 '''Note''': Please refer to [[Service Cost]] for information about the rates of the outgoing calls.&lt;br /&gt;
&lt;br /&gt;
== Account Restrictions ==&lt;br /&gt;
&lt;br /&gt;
These settings define the restrictions the system will use when you place calls to either USA, Canada or International Numbers.&lt;br /&gt;
&lt;br /&gt;
[[File:Account Restrictions.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Allow 411 dialing''': When enabled, you can call 411 directory assistance at the cost of $0.99 per call. &lt;br /&gt;
&lt;br /&gt;
 '''Note''': XXX-555-1212 is disabled within VoIP.ms. The only way to reach directory assistance is by dialing 411.&lt;br /&gt;
&lt;br /&gt;
'''Allow International Calls''': When Disabled, calls placed to destinations outside USA/Canada will be automatically rejected. If you are using [[Sub Accounts]] you need to setup this directly in the sub account.&lt;br /&gt;
&lt;br /&gt;
 '''Note''': This option is set to &amp;quot;No&amp;quot; by default for New accounts.&lt;br /&gt;
&lt;br /&gt;
'''Max. Call Time for US48/Canadian Calls''': With these setting you can choose how long the call will last when you call to US or Canada Numbers. If a call to a US or Canada Number exceeds the maximum time you have set in this option, it will be automatically hung up.&lt;br /&gt;
&lt;br /&gt;
'''Max. Call Time for International Calls''': Works exactly as the '''Max. Call Time for US48/Canadian Calls''' setting. If a call to an International Number exceeds the maximum time set, it will be automatically hung up.&lt;br /&gt;
&lt;br /&gt;
'''International Amount Restriction''': With this option you can set the maximum amount per minute for calls to international destinations. If the Per minute rate for an International destination exceeds the value you have set in this option, the call will not be connected.&lt;br /&gt;
&lt;br /&gt;
For example, let say you can set this option to $0.150 and you want to call a UK mobile number with area code 4470, the call will not go through because the per minute rate for this area code is $0.3048.&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' This settings depends on the International Routing you have choose.&lt;br /&gt;
&lt;br /&gt;
'''Allow Calls to Countries''': Here you can select Regions or Specific Countries to allow outgoing calls from your account. &lt;br /&gt;
&lt;br /&gt;
In order to allow calls to a specific country you can click the name of the region to expand the countries within it and select the desired countries only. You can select '''Select All''' so all the countries within the continent will be selected. You will also be able to search for a specific country using the '''Search Country''' field. If you call to a country not allowed in this section, the call will not be connected.&lt;br /&gt;
&lt;br /&gt;
Please make sure to only allow those destinations that you are planning to dial and avoid having unplanned destinations allowed.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:AllowCalls.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
 '''Note''': If you have international calls disabled, it will prevent calling to the international countries you have enabled in this list.&lt;br /&gt;
&lt;br /&gt;
== General ==&lt;br /&gt;
&lt;br /&gt;
These are the general settings used by the system when you make or receive calls.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:AcctSet Gen.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
'''e911 Default CallerID''': This enables you to use any CallerID you want to with your accounts but when you call E911 the number you specify here is used as your CallerID and the call is placed and completed successfully.&lt;br /&gt;
&lt;br /&gt;
'''Dialing Mode''': This setting allows you to set the way you're going to dial other numbers.&lt;br /&gt;
&lt;br /&gt;
*North America (Recommended): You can dial to countries part of the North American Numbering Plan Administration by dialing 10 or 11 digits. (with or without the 1 prefix).  You'll need to use the  00 or 011 prefix to call international numbers.&lt;br /&gt;
&lt;br /&gt;
*E164: You dial numbers by using Country Code first. You don't need to send the 00 or 011 prefix when using this mode, but it's still supported. &lt;br /&gt;
&lt;br /&gt;
'''CallerID Number''':  You can set the [[Caller ID|callerID Number]] you want to pass if you are using an [[Devices|ATA]], IP Phone or [[Softphones|Softphone]]. It is important to pass a valid [[Caller ID|caller id]] to ensure proper termination. If you have a a device capable of passing its own [[Caller ID|CallerID number]] such as a soft switch or [[PBXs|PBX]], you can leave this blank to set the [[Caller ID|CallerID Number]] on your side.&lt;br /&gt;
&lt;br /&gt;
'''Voicemail Associated to the Main Account''': You can select the mailbox associated to the main account. &lt;br /&gt;
&lt;br /&gt;
*Accessing Voice Mail: When you dial *97 from your main account, it will not prompt for the [[voicemail]] ID. It will also not prompt for the password if you have selected &amp;quot;Skip Password&amp;quot; option in the [[voicemail]] configuration. &lt;br /&gt;
&lt;br /&gt;
*Message Waiting Indicator: When there are new messages, notice of new message(s) will be sent. This will lead to different results depending on your type of adapter, soft phone or IP phone. For example, when using a  Linksys [[Devices|ATA]] adapter, it's usually a stutter tone with periodic ring, IP phones usually use a stutter dial tone when you pick up the line and are equipped with a blinking light, soft phones usually show up a Voicemail Icon in the dial display.&lt;br /&gt;
&lt;br /&gt;
 '''Note:'''This setting only affects the device registered with the main account. &lt;br /&gt;
      If you're using a [[Sub Accounts|sub account]] you need to set the '''Internal Extension Voicemail''' for that subaccount.&lt;br /&gt;
&lt;br /&gt;
'''Music On Hold''': Most IP Phones and [[Softphones|Softphones]] inform the VoIP server they are connected to when the HOLD Button is pressed. If you would like to present music to the person you place on hold, you can select it here.&amp;lt;br&amp;gt;&lt;br /&gt;
The following options for music are available:&amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
*'''No Music, Intermittent Bleep:''' Silent, but with a subtle intermittent bleep sound to let your callers know that they are still on the line while they hold.&lt;br /&gt;
*'''Away in the Tropics:''' From Hawaii to the Caribbean, these tracks deliver sounds of ukulele, steel drums, and steel guitars. &lt;br /&gt;
*'''Coffee and Sunrise:''' Uplifting without being perky, and positive without being too smiley. &lt;br /&gt;
*'''Coffee Shop Acoustic:''' Soothing, acoustic guitar tracks makes for a relaxed feel and sets a great atmosphere. &lt;br /&gt;
*'''Easy Listening:''' Smooth, casual tunes. &lt;br /&gt;
*'''Guitar Alchemy:''' Clever harmonics and progressive chord sequences to create a joyful and warming musical experience. &lt;br /&gt;
*'''Happy Endings:''' Uplifting, commercial style. Guitar, drums, ukulele, harmonica and bells. &lt;br /&gt;
*'''Light and Casual:''' Soothing and peaceful songs with a light, positive feeling. &lt;br /&gt;
*'''Orchestral Moods:''' Emotional and dramatic tales spun by violins, pianos and full orchestras. &lt;br /&gt;
*'''Piano Mix:'''  Smooth Piano &lt;br /&gt;
*'''Rock Me Easy:''' Feel-good music to create a relaxing atmosphere. &lt;br /&gt;
*'''Spa Sounds:''' Soft, slow and serene instrumentals. &lt;br /&gt;
&lt;br /&gt;
You can test the categories by dialing the following codes: &amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
'''*** 89''' Away in the Tropics &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 90''' Coffee and Sunrise &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 91''' Coffee Shop Acoustic &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 92''' Easy Listening &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 93''' Guitar Alchemy &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 94''' Happy Endings &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 95''' Light and Casual &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 96''' Orchestral Moods &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 97''' Piano Mix &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 98''' Rock Me Easy &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 99''' Spa Sounds&amp;lt;br&amp;gt;&lt;br /&gt;
'''***100''' No Music, Intermittent Bleep&amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
== Security ==&lt;br /&gt;
&lt;br /&gt;
Here you can change the password to access your Customer Portal as well your SIP/IAX passwords for your main account. You can also change the way SIP/IAX passwords are displayed on various pages of the &amp;quot;Customer Portal&amp;quot;.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accset sec.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Display SIP and IAX password(s) in Customer Portal''': If you enable this, the SIP/IAX passwords are going to be displayed in the Customer Portal. It is recommended that you leave this disabled. Enable this if only need if you forgot a SIP/IAX password for your account or one of your subaccounts. &lt;br /&gt;
 Note: You must enter your current customer portal password in order to enable this option.&lt;br /&gt;
&lt;br /&gt;
'''Customer Portal Password''': Here you can change the password used to log in your VoIP.ms Customer Portal.&lt;br /&gt;
&lt;br /&gt;
'''Main SIP/IAX Password''': Here is where you can change the password used to register your Main Account to one of the VoIP.ms servers.&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' For default the Main SIP/IAX Password is the same you use to login in your Customer Portal.&lt;br /&gt;
&lt;br /&gt;
== Inbound Settings==&lt;br /&gt;
&lt;br /&gt;
In this tab, you can set the protocol used for inbound calls and the device you're using for your main account.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Inbound.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Protocol for inbound DIDs''': Here you can select either SIP or IAX2 protocol. This will depend on the device you want to use with your Main account. SIP is the recommended protocol.&lt;br /&gt;
&lt;br /&gt;
'''Device type''': Make sure to select the correct type of device you're going to use with the main account to properly receive incoming calls in your device.&lt;br /&gt;
&lt;br /&gt;
== Notifications ==&lt;br /&gt;
&lt;br /&gt;
In this tab you can set whether to allow or not notifications in your mail when the balance in your account reach certain amount.&lt;br /&gt;
&lt;br /&gt;
[[File:Accsettings2.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Balance Threshold''': Here you can set an amount and when your balance goes below this amount you will receive an email to the address set in the option below. You can select a value between $1 up to $200, or if you like you can disable the notifications (this is not recommended).&lt;br /&gt;
&lt;br /&gt;
 '''Note''': Set this according to the monthly use you have in your account. &lt;br /&gt;
&lt;br /&gt;
'''Email''': Enter the email address in which you want to receive the notifications. Make sure to enter a proper email address. If you want to receive notifications in more than one email address, you can list the emails separated with commas (,). This will only work for the Balance Threshold&lt;br /&gt;
&lt;br /&gt;
 '''Note''': While the balance is below your threshold, you're going to keep receiving emails until you add new funds in your account or you change the balance threshold.&lt;br /&gt;
&lt;br /&gt;
== Default DID Routing ==&lt;br /&gt;
&lt;br /&gt;
These settings let you define the preconfigured Routing options that are going to be applied to each new DIDs that you order. However, you are still able to change the settings for the DID in the order page.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:DefaultDIDRouting.png|800px]]&lt;br /&gt;
&lt;br /&gt;
'''Select Plan''': Incoming Calls and Monthly Fee for the DID will be charged according to the Plan selected here. Plan prices change depending on the rate center. These prices will be shown when purchasing numbers.&lt;br /&gt;
&lt;br /&gt;
'''CallerID Name Lookup''': When activated, the system will perform a lookup in the LIBD/CNAM database to find the name matching the number of callers with a US/Canadian CallerID Number. The result of this query will be displayed with the following format &amp;quot;Caller ID name portion&amp;quot; &amp;lt;Caller ID number&amp;gt;.&lt;br /&gt;
&lt;br /&gt;
'''DID POP''': Here you can set the server in which your software/device is registered to or receiving the call from.&lt;br /&gt;
 '''Note''': Always make sure to register your software/device to the same server you set in this option to properly receive incoming calls.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:DefaultRouting.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Routing''': Here you can select the destination of the incoming calls to your DID number. You can routed directly to one account/sub account, an [[Digital Receptionist (IVR)]], [[Calling Queues]], [[Time Conditions]], [[Ring Groups]], etc.&lt;br /&gt;
&lt;br /&gt;
'''Failover Options''': This setting let you define where to redirect the call when the destination is '''Busy''', '''Unreachable''' or '''No Answer'''. These options can be displayed by clicking the '''Show Failover Options''' button&lt;br /&gt;
&lt;br /&gt;
 Note: You can change these settings later or enable additional settings in your Customer Portal&amp;gt;&amp;gt;DID Numbers&amp;gt;&amp;gt;Manage DID(s)&lt;br /&gt;
&lt;br /&gt;
== Newsletter ==&lt;br /&gt;
&lt;br /&gt;
'''Newsletter Subscription''': You can set if you want to receive news from voip.ms&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accset news.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
== Advanced ==&lt;br /&gt;
&lt;br /&gt;
These options allow you to set the allowed codecs, DTMF modes and NAT options. These settings are recommended for advanced users only.&lt;br /&gt;
&lt;br /&gt;
[[File:Accset adv.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''NAT (Network Address Translation)''': This setting should be set to Yes if you're behind a NAT, if not set to No. If you're unsure what setting means, is highly recommended that you leave it to Yes.&lt;br /&gt;
&lt;br /&gt;
'''DTMF Mode''': This allows you to select the DTMF mode that is going to be used for your account. If you set this to AUTO the RFC2833 (AVT) is going to be used and automatically switch to INBAND if the other end doesn't support RFC2833.&lt;br /&gt;
 Note: Its recommended that you select the same DTMF mode in your device. &lt;br /&gt;
&lt;br /&gt;
'''Allowed Codecs''': This setting allows you to select the codecs that you're going to use, you can choose to allow one or more codecs at the time.&lt;br /&gt;
 Note: It's recommended that you select Allow All and only change it if you have an specific reason to do so.&lt;br /&gt;
&lt;br /&gt;
[[category:guides]]&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/Account_Settings</id>
		<title>Account Settings</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/Account_Settings"/>
				<updated>2017-02-14T21:18:19Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;The Account Settings page is one of the most important pages in the Customer Portal, it will help you out as its name points out, to set all the proper ways to use your account, and these settings will affect the global settings on your account, from Device type to routing and even security of your account. &lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
__TOC__&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
== Account Routing ==&lt;br /&gt;
&lt;br /&gt;
These settings define the routes the system will use when you place calls to either US, Canada, Toll Free or International Numbers.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accsettings1.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Canada Routing''': You can choose between the [[Value vs Premium|'''Value''' or '''Premium''']] Route. If you are looking for the best wholesale price without any volume commitment you can use the '''Value''' route, however the CallerID number is not always guaranteed to work in this route.&lt;br /&gt;
The '''Premium''' route has a better level of quality, at a price that is a little higher than the value option. The CallerID Number is guaranteed to work for this route. Please note that the '''Canada Route''' can also be set independently per sub account, under the [[Sub Accounts]] settings. &lt;br /&gt;
&lt;br /&gt;
  Please note the US route is only available on the Premium route with a rate of $0.009 per minute.&lt;br /&gt;
&lt;br /&gt;
'''International Routing''': You can choose also the [[Value vs Premium|'''Value''' or '''Premium''']] route. Both routes are reliable, however you're going to find better quality if you're using the '''Premium''' route. The CallerID Number is not guaranteed to work for international calls even if you use the '''Premium''' route.&lt;br /&gt;
&lt;br /&gt;
'''Toll Free Routing''': This define the route the system will use when you place a Toll-Free call to Canada or USA. You can choose between the '''Value''' route (free to use) or the '''Premium''' (at 1.06 cent per minute). As the other settings, the '''Premium''' route give you better quality and the CallerID Number is guaranteed to work.&lt;br /&gt;
&lt;br /&gt;
'''Toll Free Carrier''': This setting defines the type of carrier that will be used when you dial a Toll-Free number. You can choose between '''Server Default''', '''American Carrier''' or '''Canadian Carrier'''. &lt;br /&gt;
&lt;br /&gt;
 '''Server Default:''' The Carrier for the termination of your toll-free call will be chosen depending on the geographical location of the server &lt;br /&gt;
   you are registering or connecting to. American Servers will dial through an American Carrier and Canadian servers will use a Canadian Toll-Free carrier.&lt;br /&gt;
 '''American Carrier:''' Your Toll-Free calls will always go through an American Carrier, even if you use a Canadian server. &lt;br /&gt;
 '''Canadian Carrier:''' Your Toll-Free calls will always go through a Canadian Carrier, even if you use an American server.&lt;br /&gt;
&lt;br /&gt;
For example, this feature can be useful for west based Canadian customers who wish to use the Seattle server but have their toll-free termination calls go through a Canadian carrier.&lt;br /&gt;
&lt;br /&gt;
 '''Note''': Please refer to [[Service Cost]] for information about the rates of the outgoing calls.&lt;br /&gt;
&lt;br /&gt;
== Account Restrictions ==&lt;br /&gt;
&lt;br /&gt;
These settings define the restrictions the system will use when you place calls to either USA, Canada or International Numbers.&lt;br /&gt;
&lt;br /&gt;
[[File:Account Restrictions.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Allow 411 dialing''': When enabled, you can call 411 directory assistance at the cost of $0.99 per call. &lt;br /&gt;
&lt;br /&gt;
 '''Note''': XXX-555-1212 is disabled within VoIP.ms. The only way to reach directory assistance is by dialing 411.&lt;br /&gt;
&lt;br /&gt;
'''Allow International Calls''': When Disabled, calls placed to destinations outside USA/Canada will be automatically rejected. If you are using [[Sub Accounts]] you need to setup this directly in the sub account.&lt;br /&gt;
&lt;br /&gt;
 '''Note''': This option is set to &amp;quot;No&amp;quot; by default for New accounts.&lt;br /&gt;
&lt;br /&gt;
'''Max. Call Time for US48/Canadian Calls''': With these setting you can choose how long the call will last when you call to US or Canada Numbers. If a call to a US or Canada Number exceeds the maximum time you have set in this option, it will be automatically hung up.&lt;br /&gt;
&lt;br /&gt;
'''Max. Call Time for International Calls''': Works exactly as the '''Max. Call Time for US48/Canadian Calls''' setting. If a call to an International Number exceeds the maximum time set, it will be automatically hung up.&lt;br /&gt;
&lt;br /&gt;
'''International Amount Restriction''': With this option you can set the maximum amount per minute for calls to international destinations. If the Per minute rate for an International destination exceeds the value you have set in this option, the call will not be connected.&lt;br /&gt;
&lt;br /&gt;
For example, let say you can set this option to $0.150 and you want to call a UK mobile number with area code 4470, the call will not go through because the per minute rate for this area code is $0.3048.&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' This settings depends on the International Routing you have choose.&lt;br /&gt;
&lt;br /&gt;
'''Allow Calls to Countries''': Here you can select Regions or Specific Countries to allow outgoing calls from your account. &lt;br /&gt;
&lt;br /&gt;
In order to allow calls to a specific country you can click the name of the region to expand the countries within it and select the desired countries only. You can select '''Select All''' so all the countries within the continent will be selected. You will also be able to search for a specific country using the '''Search Country''' field. If you call to a country not allowed in this section, the call will not be connected.&lt;br /&gt;
&lt;br /&gt;
Please make sure to only allow those destinations that you are planning to dial and avoid having unplanned destinations allowed.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:AllowCalls.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
 '''Note''': If you have international calls disabled, it will prevent calling to the international countries you have enabled in this list.&lt;br /&gt;
&lt;br /&gt;
== General ==&lt;br /&gt;
&lt;br /&gt;
These are the general settings used by the system when you make or receive calls.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:AcctSet Gen.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
'''e911 Default CallerID''': This enables you to use any CallerID you want to with your accounts but when you call E911 the number you specify here is used as your CallerID and the call is placed and completed successfully.&lt;br /&gt;
&lt;br /&gt;
'''Dialing Mode''': This setting allows you to set the way you're going to dial other numbers.&lt;br /&gt;
&lt;br /&gt;
*North America (Recommended): You can dial to countries part of the North American Numbering Plan Administration by dialing 10 or 11 digits. (with or without the 1 prefix).  You'll need to use the  00 or 011 prefix to call international numbers.&lt;br /&gt;
&lt;br /&gt;
*E164: You dial numbers by using Country Code first. You don't need to send the 00 or 011 prefix when using this mode, but it's still supported. &lt;br /&gt;
&lt;br /&gt;
'''CallerID Number''':  You can set the [[Caller ID|callerID Number]] you want to pass if you are using an [[Devices|ATA]], IP Phone or [[Softphones|Softphone]]. It is important to pass a valid [[Caller ID|caller id]] to ensure proper termination. If you have a a device capable of passing its own [[Caller ID|CallerID number]] such as a soft switch or [[PBXs|PBX]], you can leave this blank to set the [[Caller ID|CallerID Number]] on your side.&lt;br /&gt;
&lt;br /&gt;
'''Voicemail Associated to the Main Account''': You can select the mailbox associated to the main account. &lt;br /&gt;
&lt;br /&gt;
*Accessing Voice Mail: When you dial *97 from your main account, it will not prompt for the [[voicemail]] ID. It will also not prompt for the password if you have selected &amp;quot;Skip Password&amp;quot; option in the [[voicemail]] configuration. &lt;br /&gt;
&lt;br /&gt;
*Message Waiting Indicator: When there are new messages, notice of new message(s) will be sent. This will lead to different results depending on your type of adapter, soft phone or IP phone. For example, when using a  Linksys [[Devices|ATA]] adapter, it's usually a stutter tone with periodic ring, IP phones usually use a stutter dial tone when you pick up the line and are equipped with a blinking light, soft phones usually show up a Voicemail Icon in the dial display.&lt;br /&gt;
&lt;br /&gt;
 '''Note:'''This setting only affects the device registered with the main account. &lt;br /&gt;
      If you're using a [[Sub Accounts|sub account]] you need to set the '''Internal Extension Voicemail''' for that subaccount.&lt;br /&gt;
&lt;br /&gt;
'''Music On Hold''': Most IP Phones and [[Softphones|Softphones]] inform the VoIP server they are connected to when the HOLD Button is pressed. If you would like to present music to the person you place on hold, you can select it here.&amp;lt;br&amp;gt;&lt;br /&gt;
The following options for music are available:&amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
*'''No Music, Intermittent Bleep:''' Silent, but with a subtle intermittent bleep sound to let your callers know that they are still on the line while they hold.&lt;br /&gt;
*'''Away in the Tropics:''' From Hawaii to the Caribbean, these tracks deliver sounds of ukulele, steel drums, and steel guitars. &lt;br /&gt;
*'''Coffee and Sunrise:''' Uplifting without being perky, and positive without being too smiley. &lt;br /&gt;
*'''Coffee Shop Acoustic:''' Soothing, acoustic guitar tracks makes for a relaxed feel and sets a great atmosphere. &lt;br /&gt;
*'''Easy Listening:''' Smooth, casual tunes. &lt;br /&gt;
*'''Guitar Alchemy:''' Clever harmonics and progressive chord sequences to create a joyful and warming musical experience. &lt;br /&gt;
*'''Happy Endings:''' Uplifting, commercial style. Guitar, drums, ukulele, harmonica and bells. &lt;br /&gt;
*'''Light and Casual:''' Soothing and peaceful songs with a light, positive feeling. &lt;br /&gt;
*'''Orchestral Moods:''' Emotional and dramatic tales spun by violins, pianos and full orchestras. &lt;br /&gt;
*'''Piano Mix:'''  Smooth Piano &lt;br /&gt;
*'''Rock Me Easy:''' Feel-good music to create a relaxing atmosphere. &lt;br /&gt;
*'''Spa Sounds:''' Soft, slow and serene instrumentals. &lt;br /&gt;
&lt;br /&gt;
You can test the categories by dialing the following codes: &amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
'''*** 89''' Away in the Tropics &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 90''' Coffee and Sunrise &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 91''' Coffee Shop Acoustic &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 92''' Easy Listening &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 93''' Guitar Alchemy &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 94''' Happy Endings &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 95''' Light and Casual &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 96''' Orchestral Moods &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 97''' Piano Mix &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 98''' Rock Me Easy &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 99''' Spa Sounds&amp;lt;br&amp;gt;&lt;br /&gt;
'''***100''' No Music, Intermittent Bleep&amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
== Security ==&lt;br /&gt;
&lt;br /&gt;
Here you can change the password to access your Customer Portal as well your SIP/IAX passwords for your main account. You can also change the way SIP/IAX passwords are displayed on various pages of the &amp;quot;Customer Portal&amp;quot;.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accset sec.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Display SIP and IAX password(s) in Customer Portal''': If you enable this, the SIP/IAX passwords are going to be displayed in the Customer Portal. It is recommended that you leave this disabled. Enable this if only need if you forgot a SIP/IAX password for your account or one of your subaccounts. &lt;br /&gt;
 Note: You must enter your current customer portal password in order to enable this option.&lt;br /&gt;
&lt;br /&gt;
'''Customer Portal Password''': Here you can change the password used to log in your VoIP.ms Customer Portal.&lt;br /&gt;
&lt;br /&gt;
'''Main SIP/IAX Password''': Here is where you can change the password used to register your Main Account to one of the VoIP.ms servers.&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' For default the Main SIP/IAX Password is the same you use to login in your Customer Portal.&lt;br /&gt;
&lt;br /&gt;
== Inbound Settings==&lt;br /&gt;
&lt;br /&gt;
In this tab, you can set the protocol used for inbound calls and the device you're using for your main account.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Inbound.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Protocol for inbound DIDs''': Here you can select either SIP or IAX2 protocol. This will depend on the device you want to use with your Main account. SIP is the recommended protocol.&lt;br /&gt;
&lt;br /&gt;
'''Device type''': Make sure to select the correct type of device you're going to use with the main account to properly receive incoming calls in your device.&lt;br /&gt;
&lt;br /&gt;
== Notifications ==&lt;br /&gt;
&lt;br /&gt;
In this tab you can set whether to allow or not notifications in your mail when the balance in your account reach certain amount.&lt;br /&gt;
&lt;br /&gt;
[[File:Accsettings2.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Global Notifications Settings''': If you enable any notification and leave their email&lt;br /&gt;
                                                    information empty, then notifications will be sent to the&lt;br /&gt;
                                                    email provided in the Global Notifications Settings&lt;br /&gt;
                                                    &amp;lt;br&amp;gt;&amp;lt;br&amp;gt;&lt;br /&gt;
                                                    Please note that the &amp;amp;quot;[]Select All&amp;amp;quot; only selects or deselects&lt;br /&gt;
                                                    all the &amp;amp;quot;[]Send Notification&amp;amp;quot; boxes. &lt;br /&gt;
                                                    &amp;lt;br&amp;gt;&amp;lt;br&amp;gt;&lt;br /&gt;
                                                    The [Apply] button for &amp;amp;quot;Global Notifications Settings&amp;amp;quot; only saves the &lt;br /&gt;
                                                    &amp;amp;quot;[]Send Notification&amp;amp;quot; changes if all of them are selected or &lt;br /&gt;
                                                    all of them are deselected.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Balance Threshold''': Here you can set an amount and when your balance goes below this amount you will receive an email to the address set in the option below. You can select a value between $1 up to $200, or if you like you can disable the notifications (this is not recommended).&lt;br /&gt;
&lt;br /&gt;
 '''Note''': Set this according to the monthly use you have in your account. &lt;br /&gt;
&lt;br /&gt;
'''Email''': Enter the email address in which you want to receive the notifications. Make sure to enter a proper email address. If you want to receive notifications in more than one email address, you can list the emails separated with commas (,).&lt;br /&gt;
&lt;br /&gt;
 '''Note''': While the balance is below your threshold, you're going to keep receiving emails until you add new funds in your account or you change the balance threshold.&lt;br /&gt;
&lt;br /&gt;
== Default DID Routing ==&lt;br /&gt;
&lt;br /&gt;
These settings let you define the preconfigured Routing options that are going to be applied to each new DIDs that you order. However, you are still able to change the settings for the DID in the order page.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:DefaultDIDRouting.png|800px]]&lt;br /&gt;
&lt;br /&gt;
'''Select Plan''': Incoming Calls and Monthly Fee for the DID will be charged according to the Plan selected here. Plan prices change depending on the rate center. These prices will be shown when purchasing numbers.&lt;br /&gt;
&lt;br /&gt;
'''CallerID Name Lookup''': When activated, the system will perform a lookup in the LIBD/CNAM database to find the name matching the number of callers with a US/Canadian CallerID Number. The result of this query will be displayed with the following format &amp;quot;Caller ID name portion&amp;quot; &amp;lt;Caller ID number&amp;gt;.&lt;br /&gt;
&lt;br /&gt;
'''DID POP''': Here you can set the server in which your software/device is registered to or receiving the call from.&lt;br /&gt;
 '''Note''': Always make sure to register your software/device to the same server you set in this option to properly receive incoming calls.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:DefaultRouting.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Routing''': Here you can select the destination of the incoming calls to your DID number. You can routed directly to one account/sub account, an [[Digital Receptionist (IVR)]], [[Calling Queues]], [[Time Conditions]], [[Ring Groups]], etc.&lt;br /&gt;
&lt;br /&gt;
'''Failover Options''': This setting let you define where to redirect the call when the destination is '''Busy''', '''Unreachable''' or '''No Answer'''. These options can be displayed by clicking the '''Show Failover Options''' button&lt;br /&gt;
&lt;br /&gt;
 Note: You can change these settings later or enable additional settings in your Customer Portal&amp;gt;&amp;gt;DID Numbers&amp;gt;&amp;gt;Manage DID(s)&lt;br /&gt;
&lt;br /&gt;
== Newsletter ==&lt;br /&gt;
&lt;br /&gt;
'''Newsletter Subscription''': You can set if you want to receive news from voip.ms&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accset news.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
== Advanced ==&lt;br /&gt;
&lt;br /&gt;
These options allow you to set the allowed codecs, DTMF modes and NAT options. These settings are recommended for advanced users only.&lt;br /&gt;
&lt;br /&gt;
[[File:Accset adv.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''NAT (Network Address Translation)''': This setting should be set to Yes if you're behind a NAT, if not set to No. If you're unsure what setting means, is highly recommended that you leave it to Yes.&lt;br /&gt;
&lt;br /&gt;
'''DTMF Mode''': This allows you to select the DTMF mode that is going to be used for your account. If you set this to AUTO the RFC2833 (AVT) is going to be used and automatically switch to INBAND if the other end doesn't support RFC2833.&lt;br /&gt;
 Note: Its recommended that you select the same DTMF mode in your device. &lt;br /&gt;
&lt;br /&gt;
'''Allowed Codecs''': This setting allows you to select the codecs that you're going to use, you can choose to allow one or more codecs at the time.&lt;br /&gt;
 Note: It's recommended that you select Allow All and only change it if you have an specific reason to do so.&lt;br /&gt;
&lt;br /&gt;
[[category:guides]]&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/Account_Settings</id>
		<title>Account Settings</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/Account_Settings"/>
				<updated>2017-02-14T21:10:17Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;The Account Settings page is one of the most important pages in the Customer Portal, it will help you out as its name points out, to set all the proper ways to use your account, and these settings will affect the global settings on your account, from Device type to routing and even security of your account. &lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
__TOC__&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
== Account Routing ==&lt;br /&gt;
&lt;br /&gt;
These settings define the routes the system will use when you place calls to either US, Canada, Toll Free or International Numbers.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accsettings1.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Canada Routing''': You can choose between the [[Value vs Premium|'''Value''' or '''Premium''']] Route. If you are looking for the best wholesale price without any volume commitment you can use the '''Value''' route, however the CallerID number is not always guaranteed to work in this route.&lt;br /&gt;
The '''Premium''' route has a better level of quality, at a price that is a little higher than the value option. The CallerID Number is guaranteed to work for this route. Please note that the '''Canada Route''' can also be set independently per sub account, under the [[Sub Accounts]] settings. &lt;br /&gt;
&lt;br /&gt;
  Please note the US route is only available on the Premium route with a rate of $0.009 per minute.&lt;br /&gt;
&lt;br /&gt;
'''International Routing''': You can choose also the [[Value vs Premium|'''Value''' or '''Premium''']] route. Both routes are reliable, however you're going to find better quality if you're using the '''Premium''' route. The CallerID Number is not guaranteed to work for international calls even if you use the '''Premium''' route.&lt;br /&gt;
&lt;br /&gt;
'''Toll Free Routing''': This define the route the system will use when you place a Toll-Free call to Canada or USA. You can choose between the '''Value''' route (free to use) or the '''Premium''' (at 1.06 cent per minute). As the other settings, the '''Premium''' route give you better quality and the CallerID Number is guaranteed to work.&lt;br /&gt;
&lt;br /&gt;
'''Toll Free Carrier''': This setting defines the type of carrier that will be used when you dial a Toll-Free number. You can choose between '''Server Default''', '''American Carrier''' or '''Canadian Carrier'''. &lt;br /&gt;
&lt;br /&gt;
 '''Server Default:''' The Carrier for the termination of your toll-free call will be chosen depending on the geographical location of the server &lt;br /&gt;
   you are registering or connecting to. American Servers will dial through an American Carrier and Canadian servers will use a Canadian Toll-Free carrier.&lt;br /&gt;
 '''American Carrier:''' Your Toll-Free calls will always go through an American Carrier, even if you use a Canadian server. &lt;br /&gt;
 '''Canadian Carrier:''' Your Toll-Free calls will always go through a Canadian Carrier, even if you use an American server.&lt;br /&gt;
&lt;br /&gt;
For example, this feature can be useful for west based Canadian customers who wish to use the Seattle server but have their toll-free termination calls go through a Canadian carrier.&lt;br /&gt;
&lt;br /&gt;
 '''Note''': Please refer to [[Service Cost]] for information about the rates of the outgoing calls.&lt;br /&gt;
&lt;br /&gt;
== Account Restrictions ==&lt;br /&gt;
&lt;br /&gt;
These settings define the restrictions the system will use when you place calls to either USA, Canada or International Numbers.&lt;br /&gt;
&lt;br /&gt;
[[File:Account Restrictions.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Allow 411 dialing''': When enabled, you can call 411 directory assistance at the cost of $0.99 per call. &lt;br /&gt;
&lt;br /&gt;
 '''Note''': XXX-555-1212 is disabled within VoIP.ms. The only way to reach directory assistance is by dialing 411.&lt;br /&gt;
&lt;br /&gt;
'''Allow International Calls''': When Disabled, calls placed to destinations outside USA/Canada will be automatically rejected. If you are using [[Sub Accounts]] you need to setup this directly in the sub account.&lt;br /&gt;
&lt;br /&gt;
 '''Note''': This option is set to &amp;quot;No&amp;quot; by default for New accounts.&lt;br /&gt;
&lt;br /&gt;
'''Max. Call Time for US48/Canadian Calls''': With these setting you can choose how long the call will last when you call to US or Canada Numbers. If a call to a US or Canada Number exceeds the maximum time you have set in this option, it will be automatically hung up.&lt;br /&gt;
&lt;br /&gt;
'''Max. Call Time for International Calls''': Works exactly as the '''Max. Call Time for US48/Canadian Calls''' setting. If a call to an International Number exceeds the maximum time set, it will be automatically hung up.&lt;br /&gt;
&lt;br /&gt;
'''International Amount Restriction''': With this option you can set the maximum amount per minute for calls to international destinations. If the Per minute rate for an International destination exceeds the value you have set in this option, the call will not be connected.&lt;br /&gt;
&lt;br /&gt;
For example, let say you can set this option to $0.150 and you want to call a UK mobile number with area code 4470, the call will not go through because the per minute rate for this area code is $0.3048.&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' This settings depends on the International Routing you have choose.&lt;br /&gt;
&lt;br /&gt;
'''Allow Calls to Countries''': Here you can select Regions or Specific Countries to allow outgoing calls from your account. &lt;br /&gt;
&lt;br /&gt;
In order to allow calls to a specific country you can click the name of the region to expand the countries within it and select the desired countries only. You can select '''Select All''' so all the countries within the continent will be selected. You will also be able to search for a specific country using the '''Search Country''' field. If you call to a country not allowed in this section, the call will not be connected.&lt;br /&gt;
&lt;br /&gt;
Please make sure to only allow those destinations that you are planning to dial and avoid having unplanned destinations allowed.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:AllowCalls.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
 '''Note''': If you have international calls disabled, it will prevent calling to the international countries you have enabled in this list.&lt;br /&gt;
&lt;br /&gt;
== General ==&lt;br /&gt;
&lt;br /&gt;
These are the general settings used by the system when you make or receive calls.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:AcctSet Gen.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
'''e911 Default CallerID''': This enables you to use any CallerID you want to with your accounts but when you call E911 the number you specify here is used as your CallerID and the call is placed and completed successfully.&lt;br /&gt;
&lt;br /&gt;
'''Dialing Mode''': This setting allows you to set the way you're going to dial other numbers.&lt;br /&gt;
&lt;br /&gt;
*North America (Recommended): You can dial to countries part of the North American Numbering Plan Administration by dialing 10 or 11 digits. (with or without the 1 prefix).  You'll need to use the  00 or 011 prefix to call international numbers.&lt;br /&gt;
&lt;br /&gt;
*E164: You dial numbers by using Country Code first. You don't need to send the 00 or 011 prefix when using this mode, but it's still supported. &lt;br /&gt;
&lt;br /&gt;
'''CallerID Number''':  You can set the [[Caller ID|callerID Number]] you want to pass if you are using an [[Devices|ATA]], IP Phone or [[Softphones|Softphone]]. It is important to pass a valid [[Caller ID|caller id]] to ensure proper termination. If you have a a device capable of passing its own [[Caller ID|CallerID number]] such as a soft switch or [[PBXs|PBX]], you can leave this blank to set the [[Caller ID|CallerID Number]] on your side.&lt;br /&gt;
&lt;br /&gt;
'''Voicemail Associated to the Main Account''': You can select the mailbox associated to the main account. &lt;br /&gt;
&lt;br /&gt;
*Accessing Voice Mail: When you dial *97 from your main account, it will not prompt for the [[voicemail]] ID. It will also not prompt for the password if you have selected &amp;quot;Skip Password&amp;quot; option in the [[voicemail]] configuration. &lt;br /&gt;
&lt;br /&gt;
*Message Waiting Indicator: When there are new messages, notice of new message(s) will be sent. This will lead to different results depending on your type of adapter, soft phone or IP phone. For example, when using a  Linksys [[Devices|ATA]] adapter, it's usually a stutter tone with periodic ring, IP phones usually use a stutter dial tone when you pick up the line and are equipped with a blinking light, soft phones usually show up a Voicemail Icon in the dial display.&lt;br /&gt;
&lt;br /&gt;
 '''Note:'''This setting only affects the device registered with the main account. &lt;br /&gt;
      If you're using a [[Sub Accounts|sub account]] you need to set the '''Internal Extension Voicemail''' for that subaccount.&lt;br /&gt;
&lt;br /&gt;
'''Music On Hold''': Most IP Phones and [[Softphones|Softphones]] inform the VoIP server they are connected to when the HOLD Button is pressed. If you would like to present music to the person you place on hold, you can select it here.&amp;lt;br&amp;gt;&lt;br /&gt;
The following options for music are available:&amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
*'''No Music, Intermittent Bleep:''' Silent, but with a subtle intermittent bleep sound to let your callers know that they are still on the line while they hold.&lt;br /&gt;
*'''Away in the Tropics:''' From Hawaii to the Caribbean, these tracks deliver sounds of ukulele, steel drums, and steel guitars. &lt;br /&gt;
*'''Coffee and Sunrise:''' Uplifting without being perky, and positive without being too smiley. &lt;br /&gt;
*'''Coffee Shop Acoustic:''' Soothing, acoustic guitar tracks makes for a relaxed feel and sets a great atmosphere. &lt;br /&gt;
*'''Easy Listening:''' Smooth, casual tunes. &lt;br /&gt;
*'''Guitar Alchemy:''' Clever harmonics and progressive chord sequences to create a joyful and warming musical experience. &lt;br /&gt;
*'''Happy Endings:''' Uplifting, commercial style. Guitar, drums, ukulele, harmonica and bells. &lt;br /&gt;
*'''Light and Casual:''' Soothing and peaceful songs with a light, positive feeling. &lt;br /&gt;
*'''Orchestral Moods:''' Emotional and dramatic tales spun by violins, pianos and full orchestras. &lt;br /&gt;
*'''Piano Mix:'''  Smooth Piano &lt;br /&gt;
*'''Rock Me Easy:''' Feel-good music to create a relaxing atmosphere. &lt;br /&gt;
*'''Spa Sounds:''' Soft, slow and serene instrumentals. &lt;br /&gt;
&lt;br /&gt;
You can test the categories by dialing the following codes: &amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
'''*** 89''' Away in the Tropics &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 90''' Coffee and Sunrise &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 91''' Coffee Shop Acoustic &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 92''' Easy Listening &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 93''' Guitar Alchemy &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 94''' Happy Endings &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 95''' Light and Casual &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 96''' Orchestral Moods &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 97''' Piano Mix &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 98''' Rock Me Easy &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 99''' Spa Sounds&amp;lt;br&amp;gt;&lt;br /&gt;
'''***100''' No Music, Intermittent Bleep&amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
== Security ==&lt;br /&gt;
&lt;br /&gt;
Here you can change the password to access your Customer Portal as well your SIP/IAX passwords for your main account. You can also change the way SIP/IAX passwords are displayed on various pages of the &amp;quot;Customer Portal&amp;quot;.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accset sec.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Display SIP and IAX password(s) in Customer Portal''': If you enable this, the SIP/IAX passwords are going to be displayed in the Customer Portal. It is recommended that you leave this disabled. Enable this if only need if you forgot a SIP/IAX password for your account or one of your subaccounts. &lt;br /&gt;
 Note: You must enter your current customer portal password in order to enable this option.&lt;br /&gt;
&lt;br /&gt;
'''Customer Portal Password''': Here you can change the password used to log in your VoIP.ms Customer Portal.&lt;br /&gt;
&lt;br /&gt;
'''Main SIP/IAX Password''': Here is where you can change the password used to register your Main Account to one of the VoIP.ms servers.&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' For default the Main SIP/IAX Password is the same you use to login in your Customer Portal.&lt;br /&gt;
&lt;br /&gt;
== Inbound Settings==&lt;br /&gt;
&lt;br /&gt;
In this tab, you can set the protocol used for inbound calls and the device you're using for your main account.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Inbound.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Protocol for inbound DIDs''': Here you can select either SIP or IAX2 protocol. This will depend on the device you want to use with your Main account. SIP is the recommended protocol.&lt;br /&gt;
&lt;br /&gt;
'''Device type''': Make sure to select the correct type of device you're going to use with the main account to properly receive incoming calls in your device.&lt;br /&gt;
&lt;br /&gt;
== Notifications ==&lt;br /&gt;
&lt;br /&gt;
In this tab you can set whether to allow or not notifications in your mail when the balance in your account reach certain amount.&lt;br /&gt;
&lt;br /&gt;
[[File:Accsettings2.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Balance Threshold''': Here you can set an amount and when your balance goes below this amount you will receive an email to the address set in the option below. You can select a value between $1 up to $200, or if you like you can disable the notifications (this is not recommended).&lt;br /&gt;
&lt;br /&gt;
 '''Note''': Set this according to the monthly use you have in your account. &lt;br /&gt;
&lt;br /&gt;
'''Email''': Enter the email address in which you want to receive the notifications. Make sure to enter a proper email address. If you want to receive notifications in more than one email address, you can list the emails separated with commas (,).&lt;br /&gt;
&lt;br /&gt;
 '''Note''': While the balance is below your threshold, you're going to keep receiving emails until you add new funds in your account or you change the balance threshold.&lt;br /&gt;
&lt;br /&gt;
== Default DID Routing ==&lt;br /&gt;
&lt;br /&gt;
These settings let you define the preconfigured Routing options that are going to be applied to each new DIDs that you order. However, you are still able to change the settings for the DID in the order page.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:DefaultDIDRouting.png|800px]]&lt;br /&gt;
&lt;br /&gt;
'''Select Plan''': Incoming Calls and Monthly Fee for the DID will be charged according to the Plan selected here. Plan prices change depending on the rate center. These prices will be shown when purchasing numbers.&lt;br /&gt;
&lt;br /&gt;
'''CallerID Name Lookup''': When activated, the system will perform a lookup in the LIBD/CNAM database to find the name matching the number of callers with a US/Canadian CallerID Number. The result of this query will be displayed with the following format &amp;quot;Caller ID name portion&amp;quot; &amp;lt;Caller ID number&amp;gt;.&lt;br /&gt;
&lt;br /&gt;
'''DID POP''': Here you can set the server in which your software/device is registered to or receiving the call from.&lt;br /&gt;
 '''Note''': Always make sure to register your software/device to the same server you set in this option to properly receive incoming calls.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:DefaultRouting.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Routing''': Here you can select the destination of the incoming calls to your DID number. You can routed directly to one account/sub account, an [[Digital Receptionist (IVR)]], [[Calling Queues]], [[Time Conditions]], [[Ring Groups]], etc.&lt;br /&gt;
&lt;br /&gt;
'''Failover Options''': This setting let you define where to redirect the call when the destination is '''Busy''', '''Unreachable''' or '''No Answer'''. These options can be displayed by clicking the '''Show Failover Options''' button&lt;br /&gt;
&lt;br /&gt;
 Note: You can change these settings later or enable additional settings in your Customer Portal&amp;gt;&amp;gt;DID Numbers&amp;gt;&amp;gt;Manage DID(s)&lt;br /&gt;
&lt;br /&gt;
== Newsletter ==&lt;br /&gt;
&lt;br /&gt;
'''Newsletter Subscription''': You can set if you want to receive news from voip.ms&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accset news.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
== Advanced ==&lt;br /&gt;
&lt;br /&gt;
These options allow you to set the allowed codecs, DTMF modes and NAT options. These settings are recommended for advanced users only.&lt;br /&gt;
&lt;br /&gt;
[[File:Accset adv.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''NAT (Network Address Translation)''': This setting should be set to Yes if you're behind a NAT, if not set to No. If you're unsure what setting means, is highly recommended that you leave it to Yes.&lt;br /&gt;
&lt;br /&gt;
'''DTMF Mode''': This allows you to select the DTMF mode that is going to be used for your account. If you set this to AUTO the RFC2833 (AVT) is going to be used and automatically switch to INBAND if the other end doesn't support RFC2833.&lt;br /&gt;
 Note: Its recommended that you select the same DTMF mode in your device. &lt;br /&gt;
&lt;br /&gt;
'''Allowed Codecs''': This setting allows you to select the codecs that you're going to use, you can choose to allow one or more codecs at the time.&lt;br /&gt;
 Note: It's recommended that you select Allow All and only change it if you have an specific reason to do so.&lt;br /&gt;
&lt;br /&gt;
[[category:guides]]&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/Account_Settings</id>
		<title>Account Settings</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/Account_Settings"/>
				<updated>2017-02-14T21:09:37Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;The Account Settings page is one of the most important pages in the Customer Portal, it will help you out as its name points out, to set all the proper ways to use your account, and these settings will affect the global settings on your account, from Device type to routing and even security of your account. &lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
__TOC__&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
== Account Routing ==&lt;br /&gt;
&lt;br /&gt;
These settings define the routes the system will use when you place calls to either US, Canada, Toll Free or International Numbers.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accsettings1.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Canada Routing''': You can choose between the [[Value vs Premium|'''Value''' or '''Premium''']] Route. If you are looking for the best wholesale price without any volume commitment you can use the '''Value''' route, however the CallerID number is not always guaranteed to work in this route.&lt;br /&gt;
The '''Premium''' route has a better level of quality, at a price that is a little higher than the value option. The CallerID Number is guaranteed to work for this route. Please note that the '''Canada Route''' can also be set independently per sub account, under the [[Sub Accounts]] settings. &lt;br /&gt;
&lt;br /&gt;
  Please note the US route is only available on the Premium route with a rate of $0.009 per minute.&lt;br /&gt;
&lt;br /&gt;
'''International Routing''': You can choose also the [[Value vs Premium|'''Value''' or '''Premium''']] route. Both routes are reliable, however you're going to find better quality if you're using the '''Premium''' route. The CallerID Number is not guaranteed to work for international calls even if you use the '''Premium''' route.&lt;br /&gt;
&lt;br /&gt;
'''Toll Free Routing''': This define the route the system will use when you place a Toll-Free call to Canada or USA. You can choose between the '''Value''' route (free to use) or the '''Premium''' (at 1.06 cent per minute). As the other settings, the '''Premium''' route give you better quality and the CallerID Number is guaranteed to work.&lt;br /&gt;
&lt;br /&gt;
'''Toll Free Carrier''': This setting defines the type of carrier that will be used when you dial a Toll-Free number. You can choose between '''Server Default''', '''American Carrier''' or '''Canadian Carrier'''. &lt;br /&gt;
&lt;br /&gt;
 '''Server Default:''' The Carrier for the termination of your toll-free call will be chosen depending on the geographical location of the server &lt;br /&gt;
   you are registering or connecting to. American Servers will dial through an American Carrier and Canadian servers will use a Canadian Toll-Free carrier.&lt;br /&gt;
 '''American Carrier:''' Your Toll-Free calls will always go through an American Carrier, even if you use a Canadian server. &lt;br /&gt;
 '''Canadian Carrier:''' Your Toll-Free calls will always go through a Canadian Carrier, even if you use an American server.&lt;br /&gt;
&lt;br /&gt;
For example, this feature can be useful for west based Canadian customers who wish to use the Seattle server but have their toll-free termination calls go through a Canadian carrier.&lt;br /&gt;
&lt;br /&gt;
 '''Note''': Please refer to [[Service Cost]] for information about the rates of the outgoing calls.&lt;br /&gt;
&lt;br /&gt;
== Account Restrictions ==&lt;br /&gt;
&lt;br /&gt;
These settings define the restrictions the system will use when you place calls to either USA, Canada or International Numbers.&lt;br /&gt;
&lt;br /&gt;
[[File:Account Restrictions.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Allow 411 dialing''': When enabled, you can call 411 directory assistance at the cost of $0.99 per call. &lt;br /&gt;
&lt;br /&gt;
 '''Note''': XXX-555-1212 is disabled within VoIP.ms. The only way to reach directory assistance is by dialing 411.&lt;br /&gt;
&lt;br /&gt;
'''Allow International Calls''': When Disabled, calls placed to destinations outside USA/Canada will be automatically rejected. If you are using [[Sub Accounts]] you need to setup this directly in the sub account.&lt;br /&gt;
&lt;br /&gt;
 '''Note''': This option is set to &amp;quot;No&amp;quot; by default for New accounts.&lt;br /&gt;
&lt;br /&gt;
'''Max. Call Time for US48/Canadian Calls''': With these setting you can choose how long the call will last when you call to US or Canada Numbers. If a call to a US or Canada Number exceeds the maximum time you have set in this option, it will be automatically hung up.&lt;br /&gt;
&lt;br /&gt;
'''Max. Call Time for International Calls''': Works exactly as the '''Max. Call Time for US48/Canadian Calls''' setting. If a call to an International Number exceeds the maximum time set, it will be automatically hung up.&lt;br /&gt;
&lt;br /&gt;
'''International Amount Restriction''': With this option you can set the maximum amount per minute for calls to international destinations. If the Per minute rate for an International destination exceeds the value you have set in this option, the call will not be connected.&lt;br /&gt;
&lt;br /&gt;
For example, let say you can set this option to $0.150 and you want to call a UK mobile number with area code 4470, the call will not go through because the per minute rate for this area code is $0.3048.&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' This settings depends on the International Routing you have choose.&lt;br /&gt;
&lt;br /&gt;
'''Allow Calls to Countries''': Here you can select Regions or Specific Countries to allow outgoing calls from your account. &lt;br /&gt;
&lt;br /&gt;
In order to allow calls to a specific country you can click the name of the region to expand the countries within it and select the desired countries only. You can select '''Select All''' so all the countries within the continent will be selected. You will also be able to search for a specific country using the '''Search Country''' field. If you call to a country not allowed in this section, the call will not be connected.&lt;br /&gt;
&lt;br /&gt;
Please make sure to only allow those destinations that you are planning to dial and avoid having unplanned destinations allowed.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:AllowCalls.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
 '''Note''': If you have international calls disabled, it will prevent calling to the international countries you have enabled in this list.&lt;br /&gt;
&lt;br /&gt;
== General ==&lt;br /&gt;
&lt;br /&gt;
These are the general settings used by the system when you make or receive calls.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:AcctSet Gen.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
'''e911 Default CallerID''': This enables you to use any CallerID you want to with your accounts but when you call E911 the number you specify here is used as your CallerID and the call is placed and completed successfully.&lt;br /&gt;
&lt;br /&gt;
'''Dialing Mode''': This setting allows you to set the way you're going to dial other numbers.&lt;br /&gt;
&lt;br /&gt;
*North America (Recommended): You can dial to countries part of the North American Numbering Plan Administration by dialing 10 or 11 digits. (with or without the 1 prefix).  You'll need to use the  00 or 011 prefix to call international numbers.&lt;br /&gt;
&lt;br /&gt;
*E164: You dial numbers by using Country Code first. You don't need to send the 00 or 011 prefix when using this mode, but it's still supported. &lt;br /&gt;
&lt;br /&gt;
'''CallerID Number''':  You can set the [[Caller ID|callerID Number]] you want to pass if you are using an [[Devices|ATA]], IP Phone or [[Softphones|Softphone]]. It is important to pass a valid [[Caller ID|caller id]] to ensure proper termination. If you have a a device capable of passing its own [[Caller ID|CallerID number]] such as a soft switch or [[PBXs|PBX]], you can leave this blank to set the [[Caller ID|CallerID Number]] on your side.&lt;br /&gt;
&lt;br /&gt;
'''Voicemail Associated to the Main Account''': You can select the mailbox associated to the main account. &lt;br /&gt;
&lt;br /&gt;
*Accessing Voice Mail: When you dial *97 from your main account, it will not prompt for the [[voicemail]] ID. It will also not prompt for the password if you have selected &amp;quot;Skip Password&amp;quot; option in the [[voicemail]] configuration. &lt;br /&gt;
&lt;br /&gt;
*Message Waiting Indicator: When there are new messages, notice of new message(s) will be sent. This will lead to different results depending on your type of adapter, soft phone or IP phone. For example, when using a  Linksys [[Devices|ATA]] adapter, it's usually a stutter tone with periodic ring, IP phones usually use a stutter dial tone when you pick up the line and are equipped with a blinking light, soft phones usually show up a Voicemail Icon in the dial display.&lt;br /&gt;
&lt;br /&gt;
 '''Note:'''This setting only affects the device registered with the main account. &lt;br /&gt;
      If you're using a [[Sub Accounts|sub account]] you need to set the '''Internal Extension Voicemail''' for that subaccount.&lt;br /&gt;
&lt;br /&gt;
'''Music On Hold''': Most IP Phones and [[Softphones|Softphones]] inform the VoIP server they are connected to when the HOLD Button is pressed. If you would like to present music to the person you place on hold, you can select it here.&amp;lt;br&amp;gt;&lt;br /&gt;
The following options for music are available:&amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
*'''No Music, Intermittent Bleep:''' Silent, but with a subtle intermittent bleep sound to let your callers know that they are still on the line while they hold.&lt;br /&gt;
*'''Away in the Tropics:''' From Hawaii to the Caribbean, these tracks deliver sounds of ukulele, steel drums, and steel guitars. &lt;br /&gt;
*'''Coffee and Sunrise:''' Uplifting without being perky, and positive without being too smiley. &lt;br /&gt;
*'''Coffee Shop Acoustic:''' Soothing, acoustic guitar tracks makes for a relaxed feel and sets a great atmosphere. &lt;br /&gt;
*'''Easy Listening:''' Smooth, casual tunes. &lt;br /&gt;
*'''Guitar Alchemy:''' Clever harmonics and progressive chord sequences to create a joyful and warming musical experience. &lt;br /&gt;
*'''Happy Endings:''' Uplifting, commercial style. Guitar, drums, ukulele, harmonica and bells. &lt;br /&gt;
*'''Light and Casual:''' Soothing and peaceful songs with a light, positive feeling. &lt;br /&gt;
*'''Orchestral Moods:''' Emotional and dramatic tales spun by violins, pianos and full orchestras. &lt;br /&gt;
*'''Piano Mix:'''  Smooth Piano &lt;br /&gt;
*'''Rock Me Easy:''' Feel-good music to create a relaxing atmosphere. &lt;br /&gt;
*'''Spa Sounds:''' Soft, slow and serene instrumentals. &lt;br /&gt;
&lt;br /&gt;
You can test the categories by dialing the following codes: &amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
'''*** 89''' Away in the Tropics &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 90''' Coffee and Sunrise &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 91''' Coffee Shop Acoustic &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 92''' Easy Listening &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 93''' Guitar Alchemy &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 94''' Happy Endings &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 95''' Light and Casual &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 96''' Orchestral Moods &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 97''' Piano Mix &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 98''' Rock Me Easy &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 99''' Spa Sounds&amp;lt;br&amp;gt;&lt;br /&gt;
'''***100''' No Music, Intermittent Bleep&amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
== Security ==&lt;br /&gt;
&lt;br /&gt;
Here you can change the password to access your Customer Portal as well your SIP/IAX passwords for your main account. You can also change the way SIP/IAX passwords are displayed on various pages of the &amp;quot;Customer Portal&amp;quot;.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accset sec.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Display SIP and IAX password(s) in Customer Portal''': If you enable this, the SIP/IAX passwords are going to be displayed in the Customer Portal. It is recommended that you leave this disabled. Enable this if only need if you forgot a SIP/IAX password for your account or one of your subaccounts. &lt;br /&gt;
 Note: You must enter your current customer portal password in order to enable this option.&lt;br /&gt;
&lt;br /&gt;
'''Customer Portal Password''': Here you can change the password used to log in your VoIP.ms Customer Portal.&lt;br /&gt;
&lt;br /&gt;
'''Main SIP/IAX Password''': Here is where you can change the password used to register your Main Account to one of the VoIP.ms servers.&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' For default the Main SIP/IAX Password is the same you use to login in your Customer Portal.&lt;br /&gt;
&lt;br /&gt;
== Inbound Settings==&lt;br /&gt;
&lt;br /&gt;
In this tab, you can set the protocol used for inbound calls and the device you're using for your main account.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Inbound.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Protocol for inbound DIDs''': Here you can select either SIP or IAX2 protocol. This will depend on the device you want to use with your Main account. SIP is the recommended protocol.&lt;br /&gt;
&lt;br /&gt;
'''Device type''': Make sure to select the correct type of device you're going to use with the main account to properly receive incoming calls in your device.&lt;br /&gt;
&lt;br /&gt;
== Notifications ==&lt;br /&gt;
&lt;br /&gt;
In this tab you can set whether to allow or not notifications in your mail when the balance in your account reach certain amount.&lt;br /&gt;
&lt;br /&gt;
[[File:Accsettings2.png|800px]]&lt;br /&gt;
&lt;br /&gt;
'''Balance Threshold''': Here you can set an amount and when your balance goes below this amount you will receive an email to the address set in the option below. You can select a value between $1 up to $200, or if you like you can disable the notifications (this is not recommended).&lt;br /&gt;
&lt;br /&gt;
 '''Note''': Set this according to the monthly use you have in your account. &lt;br /&gt;
&lt;br /&gt;
'''Email''': Enter the email address in which you want to receive the notifications. Make sure to enter a proper email address. If you want to receive notifications in more than one email address, you can list the emails separated with commas (,).&lt;br /&gt;
&lt;br /&gt;
 '''Note''': While the balance is below your threshold, you're going to keep receiving emails until you add new funds in your account or you change the balance threshold.&lt;br /&gt;
&lt;br /&gt;
== Default DID Routing ==&lt;br /&gt;
&lt;br /&gt;
These settings let you define the preconfigured Routing options that are going to be applied to each new DIDs that you order. However, you are still able to change the settings for the DID in the order page.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:DefaultDIDRouting.png|800px]]&lt;br /&gt;
&lt;br /&gt;
'''Select Plan''': Incoming Calls and Monthly Fee for the DID will be charged according to the Plan selected here. Plan prices change depending on the rate center. These prices will be shown when purchasing numbers.&lt;br /&gt;
&lt;br /&gt;
'''CallerID Name Lookup''': When activated, the system will perform a lookup in the LIBD/CNAM database to find the name matching the number of callers with a US/Canadian CallerID Number. The result of this query will be displayed with the following format &amp;quot;Caller ID name portion&amp;quot; &amp;lt;Caller ID number&amp;gt;.&lt;br /&gt;
&lt;br /&gt;
'''DID POP''': Here you can set the server in which your software/device is registered to or receiving the call from.&lt;br /&gt;
 '''Note''': Always make sure to register your software/device to the same server you set in this option to properly receive incoming calls.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:DefaultRouting.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Routing''': Here you can select the destination of the incoming calls to your DID number. You can routed directly to one account/sub account, an [[Digital Receptionist (IVR)]], [[Calling Queues]], [[Time Conditions]], [[Ring Groups]], etc.&lt;br /&gt;
&lt;br /&gt;
'''Failover Options''': This setting let you define where to redirect the call when the destination is '''Busy''', '''Unreachable''' or '''No Answer'''. These options can be displayed by clicking the '''Show Failover Options''' button&lt;br /&gt;
&lt;br /&gt;
 Note: You can change these settings later or enable additional settings in your Customer Portal&amp;gt;&amp;gt;DID Numbers&amp;gt;&amp;gt;Manage DID(s)&lt;br /&gt;
&lt;br /&gt;
== Newsletter ==&lt;br /&gt;
&lt;br /&gt;
'''Newsletter Subscription''': You can set if you want to receive news from voip.ms&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accset news.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
== Advanced ==&lt;br /&gt;
&lt;br /&gt;
These options allow you to set the allowed codecs, DTMF modes and NAT options. These settings are recommended for advanced users only.&lt;br /&gt;
&lt;br /&gt;
[[File:Accset adv.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''NAT (Network Address Translation)''': This setting should be set to Yes if you're behind a NAT, if not set to No. If you're unsure what setting means, is highly recommended that you leave it to Yes.&lt;br /&gt;
&lt;br /&gt;
'''DTMF Mode''': This allows you to select the DTMF mode that is going to be used for your account. If you set this to AUTO the RFC2833 (AVT) is going to be used and automatically switch to INBAND if the other end doesn't support RFC2833.&lt;br /&gt;
 Note: Its recommended that you select the same DTMF mode in your device. &lt;br /&gt;
&lt;br /&gt;
'''Allowed Codecs''': This setting allows you to select the codecs that you're going to use, you can choose to allow one or more codecs at the time.&lt;br /&gt;
 Note: It's recommended that you select Allow All and only change it if you have an specific reason to do so.&lt;br /&gt;
&lt;br /&gt;
[[category:guides]]&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/Account_Settings</id>
		<title>Account Settings</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/Account_Settings"/>
				<updated>2017-02-14T21:09:01Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;The Account Settings page is one of the most important pages in the Customer Portal, it will help you out as its name points out, to set all the proper ways to use your account, and these settings will affect the global settings on your account, from Device type to routing and even security of your account. &lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
__TOC__&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
== Account Routing ==&lt;br /&gt;
&lt;br /&gt;
These settings define the routes the system will use when you place calls to either US, Canada, Toll Free or International Numbers.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accsettings1.png]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Canada Routing''': You can choose between the [[Value vs Premium|'''Value''' or '''Premium''']] Route. If you are looking for the best wholesale price without any volume commitment you can use the '''Value''' route, however the CallerID number is not always guaranteed to work in this route.&lt;br /&gt;
The '''Premium''' route has a better level of quality, at a price that is a little higher than the value option. The CallerID Number is guaranteed to work for this route. Please note that the '''Canada Route''' can also be set independently per sub account, under the [[Sub Accounts]] settings. &lt;br /&gt;
&lt;br /&gt;
  Please note the US route is only available on the Premium route with a rate of $0.009 per minute.&lt;br /&gt;
&lt;br /&gt;
'''International Routing''': You can choose also the [[Value vs Premium|'''Value''' or '''Premium''']] route. Both routes are reliable, however you're going to find better quality if you're using the '''Premium''' route. The CallerID Number is not guaranteed to work for international calls even if you use the '''Premium''' route.&lt;br /&gt;
&lt;br /&gt;
'''Toll Free Routing''': This define the route the system will use when you place a Toll-Free call to Canada or USA. You can choose between the '''Value''' route (free to use) or the '''Premium''' (at 1.06 cent per minute). As the other settings, the '''Premium''' route give you better quality and the CallerID Number is guaranteed to work.&lt;br /&gt;
&lt;br /&gt;
'''Toll Free Carrier''': This setting defines the type of carrier that will be used when you dial a Toll-Free number. You can choose between '''Server Default''', '''American Carrier''' or '''Canadian Carrier'''. &lt;br /&gt;
&lt;br /&gt;
 '''Server Default:''' The Carrier for the termination of your toll-free call will be chosen depending on the geographical location of the server &lt;br /&gt;
   you are registering or connecting to. American Servers will dial through an American Carrier and Canadian servers will use a Canadian Toll-Free carrier.&lt;br /&gt;
 '''American Carrier:''' Your Toll-Free calls will always go through an American Carrier, even if you use a Canadian server. &lt;br /&gt;
 '''Canadian Carrier:''' Your Toll-Free calls will always go through a Canadian Carrier, even if you use an American server.&lt;br /&gt;
&lt;br /&gt;
For example, this feature can be useful for west based Canadian customers who wish to use the Seattle server but have their toll-free termination calls go through a Canadian carrier.&lt;br /&gt;
&lt;br /&gt;
 '''Note''': Please refer to [[Service Cost]] for information about the rates of the outgoing calls.&lt;br /&gt;
&lt;br /&gt;
== Account Restrictions ==&lt;br /&gt;
&lt;br /&gt;
These settings define the restrictions the system will use when you place calls to either USA, Canada or International Numbers.&lt;br /&gt;
&lt;br /&gt;
[[File:Account Restrictions.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Allow 411 dialing''': When enabled, you can call 411 directory assistance at the cost of $0.99 per call. &lt;br /&gt;
&lt;br /&gt;
 '''Note''': XXX-555-1212 is disabled within VoIP.ms. The only way to reach directory assistance is by dialing 411.&lt;br /&gt;
&lt;br /&gt;
'''Allow International Calls''': When Disabled, calls placed to destinations outside USA/Canada will be automatically rejected. If you are using [[Sub Accounts]] you need to setup this directly in the sub account.&lt;br /&gt;
&lt;br /&gt;
 '''Note''': This option is set to &amp;quot;No&amp;quot; by default for New accounts.&lt;br /&gt;
&lt;br /&gt;
'''Max. Call Time for US48/Canadian Calls''': With these setting you can choose how long the call will last when you call to US or Canada Numbers. If a call to a US or Canada Number exceeds the maximum time you have set in this option, it will be automatically hung up.&lt;br /&gt;
&lt;br /&gt;
'''Max. Call Time for International Calls''': Works exactly as the '''Max. Call Time for US48/Canadian Calls''' setting. If a call to an International Number exceeds the maximum time set, it will be automatically hung up.&lt;br /&gt;
&lt;br /&gt;
'''International Amount Restriction''': With this option you can set the maximum amount per minute for calls to international destinations. If the Per minute rate for an International destination exceeds the value you have set in this option, the call will not be connected.&lt;br /&gt;
&lt;br /&gt;
For example, let say you can set this option to $0.150 and you want to call a UK mobile number with area code 4470, the call will not go through because the per minute rate for this area code is $0.3048.&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' This settings depends on the International Routing you have choose.&lt;br /&gt;
&lt;br /&gt;
'''Allow Calls to Countries''': Here you can select Regions or Specific Countries to allow outgoing calls from your account. &lt;br /&gt;
&lt;br /&gt;
In order to allow calls to a specific country you can click the name of the region to expand the countries within it and select the desired countries only. You can select '''Select All''' so all the countries within the continent will be selected. You will also be able to search for a specific country using the '''Search Country''' field. If you call to a country not allowed in this section, the call will not be connected.&lt;br /&gt;
&lt;br /&gt;
Please make sure to only allow those destinations that you are planning to dial and avoid having unplanned destinations allowed.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:AllowCalls.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
 '''Note''': If you have international calls disabled, it will prevent calling to the international countries you have enabled in this list.&lt;br /&gt;
&lt;br /&gt;
== General ==&lt;br /&gt;
&lt;br /&gt;
These are the general settings used by the system when you make or receive calls.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:AcctSet Gen.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
'''e911 Default CallerID''': This enables you to use any CallerID you want to with your accounts but when you call E911 the number you specify here is used as your CallerID and the call is placed and completed successfully.&lt;br /&gt;
&lt;br /&gt;
'''Dialing Mode''': This setting allows you to set the way you're going to dial other numbers.&lt;br /&gt;
&lt;br /&gt;
*North America (Recommended): You can dial to countries part of the North American Numbering Plan Administration by dialing 10 or 11 digits. (with or without the 1 prefix).  You'll need to use the  00 or 011 prefix to call international numbers.&lt;br /&gt;
&lt;br /&gt;
*E164: You dial numbers by using Country Code first. You don't need to send the 00 or 011 prefix when using this mode, but it's still supported. &lt;br /&gt;
&lt;br /&gt;
'''CallerID Number''':  You can set the [[Caller ID|callerID Number]] you want to pass if you are using an [[Devices|ATA]], IP Phone or [[Softphones|Softphone]]. It is important to pass a valid [[Caller ID|caller id]] to ensure proper termination. If you have a a device capable of passing its own [[Caller ID|CallerID number]] such as a soft switch or [[PBXs|PBX]], you can leave this blank to set the [[Caller ID|CallerID Number]] on your side.&lt;br /&gt;
&lt;br /&gt;
'''Voicemail Associated to the Main Account''': You can select the mailbox associated to the main account. &lt;br /&gt;
&lt;br /&gt;
*Accessing Voice Mail: When you dial *97 from your main account, it will not prompt for the [[voicemail]] ID. It will also not prompt for the password if you have selected &amp;quot;Skip Password&amp;quot; option in the [[voicemail]] configuration. &lt;br /&gt;
&lt;br /&gt;
*Message Waiting Indicator: When there are new messages, notice of new message(s) will be sent. This will lead to different results depending on your type of adapter, soft phone or IP phone. For example, when using a  Linksys [[Devices|ATA]] adapter, it's usually a stutter tone with periodic ring, IP phones usually use a stutter dial tone when you pick up the line and are equipped with a blinking light, soft phones usually show up a Voicemail Icon in the dial display.&lt;br /&gt;
&lt;br /&gt;
 '''Note:'''This setting only affects the device registered with the main account. &lt;br /&gt;
      If you're using a [[Sub Accounts|sub account]] you need to set the '''Internal Extension Voicemail''' for that subaccount.&lt;br /&gt;
&lt;br /&gt;
'''Music On Hold''': Most IP Phones and [[Softphones|Softphones]] inform the VoIP server they are connected to when the HOLD Button is pressed. If you would like to present music to the person you place on hold, you can select it here.&amp;lt;br&amp;gt;&lt;br /&gt;
The following options for music are available:&amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
*'''No Music, Intermittent Bleep:''' Silent, but with a subtle intermittent bleep sound to let your callers know that they are still on the line while they hold.&lt;br /&gt;
*'''Away in the Tropics:''' From Hawaii to the Caribbean, these tracks deliver sounds of ukulele, steel drums, and steel guitars. &lt;br /&gt;
*'''Coffee and Sunrise:''' Uplifting without being perky, and positive without being too smiley. &lt;br /&gt;
*'''Coffee Shop Acoustic:''' Soothing, acoustic guitar tracks makes for a relaxed feel and sets a great atmosphere. &lt;br /&gt;
*'''Easy Listening:''' Smooth, casual tunes. &lt;br /&gt;
*'''Guitar Alchemy:''' Clever harmonics and progressive chord sequences to create a joyful and warming musical experience. &lt;br /&gt;
*'''Happy Endings:''' Uplifting, commercial style. Guitar, drums, ukulele, harmonica and bells. &lt;br /&gt;
*'''Light and Casual:''' Soothing and peaceful songs with a light, positive feeling. &lt;br /&gt;
*'''Orchestral Moods:''' Emotional and dramatic tales spun by violins, pianos and full orchestras. &lt;br /&gt;
*'''Piano Mix:'''  Smooth Piano &lt;br /&gt;
*'''Rock Me Easy:''' Feel-good music to create a relaxing atmosphere. &lt;br /&gt;
*'''Spa Sounds:''' Soft, slow and serene instrumentals. &lt;br /&gt;
&lt;br /&gt;
You can test the categories by dialing the following codes: &amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
'''*** 89''' Away in the Tropics &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 90''' Coffee and Sunrise &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 91''' Coffee Shop Acoustic &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 92''' Easy Listening &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 93''' Guitar Alchemy &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 94''' Happy Endings &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 95''' Light and Casual &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 96''' Orchestral Moods &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 97''' Piano Mix &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 98''' Rock Me Easy &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 99''' Spa Sounds&amp;lt;br&amp;gt;&lt;br /&gt;
'''***100''' No Music, Intermittent Bleep&amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
== Security ==&lt;br /&gt;
&lt;br /&gt;
Here you can change the password to access your Customer Portal as well your SIP/IAX passwords for your main account. You can also change the way SIP/IAX passwords are displayed on various pages of the &amp;quot;Customer Portal&amp;quot;.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accset sec.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Display SIP and IAX password(s) in Customer Portal''': If you enable this, the SIP/IAX passwords are going to be displayed in the Customer Portal. It is recommended that you leave this disabled. Enable this if only need if you forgot a SIP/IAX password for your account or one of your subaccounts. &lt;br /&gt;
 Note: You must enter your current customer portal password in order to enable this option.&lt;br /&gt;
&lt;br /&gt;
'''Customer Portal Password''': Here you can change the password used to log in your VoIP.ms Customer Portal.&lt;br /&gt;
&lt;br /&gt;
'''Main SIP/IAX Password''': Here is where you can change the password used to register your Main Account to one of the VoIP.ms servers.&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' For default the Main SIP/IAX Password is the same you use to login in your Customer Portal.&lt;br /&gt;
&lt;br /&gt;
== Inbound Settings==&lt;br /&gt;
&lt;br /&gt;
In this tab, you can set the protocol used for inbound calls and the device you're using for your main account.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Inbound.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Protocol for inbound DIDs''': Here you can select either SIP or IAX2 protocol. This will depend on the device you want to use with your Main account. SIP is the recommended protocol.&lt;br /&gt;
&lt;br /&gt;
'''Device type''': Make sure to select the correct type of device you're going to use with the main account to properly receive incoming calls in your device.&lt;br /&gt;
&lt;br /&gt;
== Notifications ==&lt;br /&gt;
&lt;br /&gt;
In this tab you can set whether to allow or not notifications in your mail when the balance in your account reach certain amount.&lt;br /&gt;
&lt;br /&gt;
[[File:Accsettings2.png|800px]]&lt;br /&gt;
&lt;br /&gt;
'''Balance Threshold''': Here you can set an amount and when your balance goes below this amount you will receive an email to the address set in the option below. You can select a value between $1 up to $200, or if you like you can disable the notifications (this is not recommended).&lt;br /&gt;
&lt;br /&gt;
 '''Note''': Set this according to the monthly use you have in your account. &lt;br /&gt;
&lt;br /&gt;
'''Email''': Enter the email address in which you want to receive the notifications. Make sure to enter a proper email address. If you want to receive notifications in more than one email address, you can list the emails separated with commas (,).&lt;br /&gt;
&lt;br /&gt;
 '''Note''': While the balance is below your threshold, you're going to keep receiving emails until you add new funds in your account or you change the balance threshold.&lt;br /&gt;
&lt;br /&gt;
== Default DID Routing ==&lt;br /&gt;
&lt;br /&gt;
These settings let you define the preconfigured Routing options that are going to be applied to each new DIDs that you order. However, you are still able to change the settings for the DID in the order page.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:DefaultDIDRouting.png|800px]]&lt;br /&gt;
&lt;br /&gt;
'''Select Plan''': Incoming Calls and Monthly Fee for the DID will be charged according to the Plan selected here. Plan prices change depending on the rate center. These prices will be shown when purchasing numbers.&lt;br /&gt;
&lt;br /&gt;
'''CallerID Name Lookup''': When activated, the system will perform a lookup in the LIBD/CNAM database to find the name matching the number of callers with a US/Canadian CallerID Number. The result of this query will be displayed with the following format &amp;quot;Caller ID name portion&amp;quot; &amp;lt;Caller ID number&amp;gt;.&lt;br /&gt;
&lt;br /&gt;
'''DID POP''': Here you can set the server in which your software/device is registered to or receiving the call from.&lt;br /&gt;
 '''Note''': Always make sure to register your software/device to the same server you set in this option to properly receive incoming calls.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:DefaultRouting.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Routing''': Here you can select the destination of the incoming calls to your DID number. You can routed directly to one account/sub account, an [[Digital Receptionist (IVR)]], [[Calling Queues]], [[Time Conditions]], [[Ring Groups]], etc.&lt;br /&gt;
&lt;br /&gt;
'''Failover Options''': This setting let you define where to redirect the call when the destination is '''Busy''', '''Unreachable''' or '''No Answer'''. These options can be displayed by clicking the '''Show Failover Options''' button&lt;br /&gt;
&lt;br /&gt;
 Note: You can change these settings later or enable additional settings in your Customer Portal&amp;gt;&amp;gt;DID Numbers&amp;gt;&amp;gt;Manage DID(s)&lt;br /&gt;
&lt;br /&gt;
== Newsletter ==&lt;br /&gt;
&lt;br /&gt;
'''Newsletter Subscription''': You can set if you want to receive news from voip.ms&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accset news.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
== Advanced ==&lt;br /&gt;
&lt;br /&gt;
These options allow you to set the allowed codecs, DTMF modes and NAT options. These settings are recommended for advanced users only.&lt;br /&gt;
&lt;br /&gt;
[[File:Accset adv.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''NAT (Network Address Translation)''': This setting should be set to Yes if you're behind a NAT, if not set to No. If you're unsure what setting means, is highly recommended that you leave it to Yes.&lt;br /&gt;
&lt;br /&gt;
'''DTMF Mode''': This allows you to select the DTMF mode that is going to be used for your account. If you set this to AUTO the RFC2833 (AVT) is going to be used and automatically switch to INBAND if the other end doesn't support RFC2833.&lt;br /&gt;
 Note: Its recommended that you select the same DTMF mode in your device. &lt;br /&gt;
&lt;br /&gt;
'''Allowed Codecs''': This setting allows you to select the codecs that you're going to use, you can choose to allow one or more codecs at the time.&lt;br /&gt;
 Note: It's recommended that you select Allow All and only change it if you have an specific reason to do so.&lt;br /&gt;
&lt;br /&gt;
[[category:guides]]&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/File:Accsettings2.png</id>
		<title>File:Accsettings2.png</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/File:Accsettings2.png"/>
				<updated>2017-02-14T21:07:48Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/Account_Settings</id>
		<title>Account Settings</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/Account_Settings"/>
				<updated>2017-02-14T21:04:12Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;The Account Settings page is one of the most important pages in the Customer Portal, it will help you out as its name points out, to set all the proper ways to use your account, and these settings will affect the global settings on your account, from Device type to routing and even security of your account. &lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
__TOC__&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
== Account Routing ==&lt;br /&gt;
&lt;br /&gt;
These settings define the routes the system will use when you place calls to either US, Canada, Toll Free or International Numbers.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accsettings1.png]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Canada Routing''': You can choose between the [[Value vs Premium|'''Value''' or '''Premium''']] Route. If you are looking for the best wholesale price without any volume commitment you can use the '''Value''' route, however the CallerID number is not always guaranteed to work in this route.&lt;br /&gt;
The '''Premium''' route has a better level of quality, at a price that is a little higher than the value option. The CallerID Number is guaranteed to work for this route. Please note that the '''Canada Route''' can also be set independently per sub account, under the [[Sub Accounts]] settings. &lt;br /&gt;
&lt;br /&gt;
  Please note the US route is only available on the Premium route with a rate of $0.009 per minute.&lt;br /&gt;
&lt;br /&gt;
'''International Routing''': You can choose also the [[Value vs Premium|'''Value''' or '''Premium''']] route. Both routes are reliable, however you're going to find better quality if you're using the '''Premium''' route. The CallerID Number is not guaranteed to work for international calls even if you use the '''Premium''' route.&lt;br /&gt;
&lt;br /&gt;
'''Toll Free Routing''': This define the route the system will use when you place a Toll-Free call to Canada or USA. You can choose between the '''Value''' route (free to use) or the '''Premium''' (at 1.06 cent per minute). As the other settings, the '''Premium''' route give you better quality and the CallerID Number is guaranteed to work.&lt;br /&gt;
&lt;br /&gt;
'''Toll Free Carrier''': This setting defines the type of carrier that will be used when you dial a Toll-Free number. You can choose between '''Server Default''', '''American Carrier''' or '''Canadian Carrier'''. &lt;br /&gt;
&lt;br /&gt;
 '''Server Default:''' The Carrier for the termination of your toll-free call will be chosen depending on the geographical location of the server &lt;br /&gt;
   you are registering or connecting to. American Servers will dial through an American Carrier and Canadian servers will use a Canadian Toll-Free carrier.&lt;br /&gt;
 '''American Carrier:''' Your Toll-Free calls will always go through an American Carrier, even if you use a Canadian server. &lt;br /&gt;
 '''Canadian Carrier:''' Your Toll-Free calls will always go through a Canadian Carrier, even if you use an American server.&lt;br /&gt;
&lt;br /&gt;
For example, this feature can be useful for west based Canadian customers who wish to use the Seattle server but have their toll-free termination calls go through a Canadian carrier.&lt;br /&gt;
&lt;br /&gt;
 '''Note''': Please refer to [[Service Cost]] for information about the rates of the outgoing calls.&lt;br /&gt;
&lt;br /&gt;
== Account Restrictions ==&lt;br /&gt;
&lt;br /&gt;
These settings define the restrictions the system will use when you place calls to either USA, Canada or International Numbers.&lt;br /&gt;
&lt;br /&gt;
[[File:Account Restrictions.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Allow 411 dialing''': When enabled, you can call 411 directory assistance at the cost of $0.99 per call. &lt;br /&gt;
&lt;br /&gt;
 '''Note''': XXX-555-1212 is disabled within VoIP.ms. The only way to reach directory assistance is by dialing 411.&lt;br /&gt;
&lt;br /&gt;
'''Allow International Calls''': When Disabled, calls placed to destinations outside USA/Canada will be automatically rejected. If you are using [[Sub Accounts]] you need to setup this directly in the sub account.&lt;br /&gt;
&lt;br /&gt;
 '''Note''': This option is set to &amp;quot;No&amp;quot; by default for New accounts.&lt;br /&gt;
&lt;br /&gt;
'''Max. Call Time for US48/Canadian Calls''': With these setting you can choose how long the call will last when you call to US or Canada Numbers. If a call to a US or Canada Number exceeds the maximum time you have set in this option, it will be automatically hung up.&lt;br /&gt;
&lt;br /&gt;
'''Max. Call Time for International Calls''': Works exactly as the '''Max. Call Time for US48/Canadian Calls''' setting. If a call to an International Number exceeds the maximum time set, it will be automatically hung up.&lt;br /&gt;
&lt;br /&gt;
'''International Amount Restriction''': With this option you can set the maximum amount per minute for calls to international destinations. If the Per minute rate for an International destination exceeds the value you have set in this option, the call will not be connected.&lt;br /&gt;
&lt;br /&gt;
For example, let say you can set this option to $0.150 and you want to call a UK mobile number with area code 4470, the call will not go through because the per minute rate for this area code is $0.3048.&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' This settings depends on the International Routing you have choose.&lt;br /&gt;
&lt;br /&gt;
'''Allow Calls to Countries''': Here you can select Regions or Specific Countries to allow outgoing calls from your account. &lt;br /&gt;
&lt;br /&gt;
In order to allow calls to a specific country you can click the name of the region to expand the countries within it and select the desired countries only. You can select '''Select All''' so all the countries within the continent will be selected. You will also be able to search for a specific country using the '''Search Country''' field. If you call to a country not allowed in this section, the call will not be connected.&lt;br /&gt;
&lt;br /&gt;
Please make sure to only allow those destinations that you are planning to dial and avoid having unplanned destinations allowed.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:AllowCalls.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
 '''Note''': If you have international calls disabled, it will prevent calling to the international countries you have enabled in this list.&lt;br /&gt;
&lt;br /&gt;
== General ==&lt;br /&gt;
&lt;br /&gt;
These are the general settings used by the system when you make or receive calls.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:AcctSet Gen.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
'''e911 Default CallerID''': This enables you to use any CallerID you want to with your accounts but when you call E911 the number you specify here is used as your CallerID and the call is placed and completed successfully.&lt;br /&gt;
&lt;br /&gt;
'''Dialing Mode''': This setting allows you to set the way you're going to dial other numbers.&lt;br /&gt;
&lt;br /&gt;
*North America (Recommended): You can dial to countries part of the North American Numbering Plan Administration by dialing 10 or 11 digits. (with or without the 1 prefix).  You'll need to use the  00 or 011 prefix to call international numbers.&lt;br /&gt;
&lt;br /&gt;
*E164: You dial numbers by using Country Code first. You don't need to send the 00 or 011 prefix when using this mode, but it's still supported. &lt;br /&gt;
&lt;br /&gt;
'''CallerID Number''':  You can set the [[Caller ID|callerID Number]] you want to pass if you are using an [[Devices|ATA]], IP Phone or [[Softphones|Softphone]]. It is important to pass a valid [[Caller ID|caller id]] to ensure proper termination. If you have a a device capable of passing its own [[Caller ID|CallerID number]] such as a soft switch or [[PBXs|PBX]], you can leave this blank to set the [[Caller ID|CallerID Number]] on your side.&lt;br /&gt;
&lt;br /&gt;
'''Voicemail Associated to the Main Account''': You can select the mailbox associated to the main account. &lt;br /&gt;
&lt;br /&gt;
*Accessing Voice Mail: When you dial *97 from your main account, it will not prompt for the [[voicemail]] ID. It will also not prompt for the password if you have selected &amp;quot;Skip Password&amp;quot; option in the [[voicemail]] configuration. &lt;br /&gt;
&lt;br /&gt;
*Message Waiting Indicator: When there are new messages, notice of new message(s) will be sent. This will lead to different results depending on your type of adapter, soft phone or IP phone. For example, when using a  Linksys [[Devices|ATA]] adapter, it's usually a stutter tone with periodic ring, IP phones usually use a stutter dial tone when you pick up the line and are equipped with a blinking light, soft phones usually show up a Voicemail Icon in the dial display.&lt;br /&gt;
&lt;br /&gt;
 '''Note:'''This setting only affects the device registered with the main account. &lt;br /&gt;
      If you're using a [[Sub Accounts|sub account]] you need to set the '''Internal Extension Voicemail''' for that subaccount.&lt;br /&gt;
&lt;br /&gt;
'''Music On Hold''': Most IP Phones and [[Softphones|Softphones]] inform the VoIP server they are connected to when the HOLD Button is pressed. If you would like to present music to the person you place on hold, you can select it here.&amp;lt;br&amp;gt;&lt;br /&gt;
The following options for music are available:&amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
*'''No Music, Intermittent Bleep:''' Silent, but with a subtle intermittent bleep sound to let your callers know that they are still on the line while they hold.&lt;br /&gt;
*'''Away in the Tropics:''' From Hawaii to the Caribbean, these tracks deliver sounds of ukulele, steel drums, and steel guitars. &lt;br /&gt;
*'''Coffee and Sunrise:''' Uplifting without being perky, and positive without being too smiley. &lt;br /&gt;
*'''Coffee Shop Acoustic:''' Soothing, acoustic guitar tracks makes for a relaxed feel and sets a great atmosphere. &lt;br /&gt;
*'''Easy Listening:''' Smooth, casual tunes. &lt;br /&gt;
*'''Guitar Alchemy:''' Clever harmonics and progressive chord sequences to create a joyful and warming musical experience. &lt;br /&gt;
*'''Happy Endings:''' Uplifting, commercial style. Guitar, drums, ukulele, harmonica and bells. &lt;br /&gt;
*'''Light and Casual:''' Soothing and peaceful songs with a light, positive feeling. &lt;br /&gt;
*'''Orchestral Moods:''' Emotional and dramatic tales spun by violins, pianos and full orchestras. &lt;br /&gt;
*'''Piano Mix:'''  Smooth Piano &lt;br /&gt;
*'''Rock Me Easy:''' Feel-good music to create a relaxing atmosphere. &lt;br /&gt;
*'''Spa Sounds:''' Soft, slow and serene instrumentals. &lt;br /&gt;
&lt;br /&gt;
You can test the categories by dialing the following codes: &amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
'''*** 89''' Away in the Tropics &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 90''' Coffee and Sunrise &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 91''' Coffee Shop Acoustic &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 92''' Easy Listening &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 93''' Guitar Alchemy &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 94''' Happy Endings &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 95''' Light and Casual &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 96''' Orchestral Moods &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 97''' Piano Mix &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 98''' Rock Me Easy &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 99''' Spa Sounds&amp;lt;br&amp;gt;&lt;br /&gt;
'''***100''' No Music, Intermittent Bleep&amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
== Security ==&lt;br /&gt;
&lt;br /&gt;
Here you can change the password to access your Customer Portal as well your SIP/IAX passwords for your main account. You can also change the way SIP/IAX passwords are displayed on various pages of the &amp;quot;Customer Portal&amp;quot;.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accset sec.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Display SIP and IAX password(s) in Customer Portal''': If you enable this, the SIP/IAX passwords are going to be displayed in the Customer Portal. It is recommended that you leave this disabled. Enable this if only need if you forgot a SIP/IAX password for your account or one of your subaccounts. &lt;br /&gt;
 Note: You must enter your current customer portal password in order to enable this option.&lt;br /&gt;
&lt;br /&gt;
'''Customer Portal Password''': Here you can change the password used to log in your VoIP.ms Customer Portal.&lt;br /&gt;
&lt;br /&gt;
'''Main SIP/IAX Password''': Here is where you can change the password used to register your Main Account to one of the VoIP.ms servers.&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' For default the Main SIP/IAX Password is the same you use to login in your Customer Portal.&lt;br /&gt;
&lt;br /&gt;
== Inbound Settings==&lt;br /&gt;
&lt;br /&gt;
In this tab, you can set the protocol used for inbound calls and the device you're using for your main account.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Inbound.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Protocol for inbound DIDs''': Here you can select either SIP or IAX2 protocol. This will depend on the device you want to use with your Main account. SIP is the recommended protocol.&lt;br /&gt;
&lt;br /&gt;
'''Device type''': Make sure to select the correct type of device you're going to use with the main account to properly receive incoming calls in your device.&lt;br /&gt;
&lt;br /&gt;
== Notifications ==&lt;br /&gt;
&lt;br /&gt;
In this tab you can set whether to allow or not notifications in your mail when the balance in your account reach certain amount.&lt;br /&gt;
&lt;br /&gt;
[[File:Accset not.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
'''Balance Threshold''': Here you can set an amount and when your balance goes below this amount you will receive an email to the address set in the option below. You can select a value between $1 up to $200, or if you like you can disable the notifications (this is not recommended).&lt;br /&gt;
&lt;br /&gt;
 '''Note''': Set this according to the monthly use you have in your account. &lt;br /&gt;
&lt;br /&gt;
'''Email''': Enter the email address in which you want to receive the notifications. Make sure to enter a proper email address. If you want to receive notifications in more than one email address, you can list the emails separated with commas (,).&lt;br /&gt;
&lt;br /&gt;
 '''Note''': While the balance is below your threshold, you're going to keep receiving emails until you add new funds in your account or you change the balance threshold.&lt;br /&gt;
&lt;br /&gt;
== Default DID Routing ==&lt;br /&gt;
&lt;br /&gt;
These settings let you define the preconfigured Routing options that are going to be applied to each new DIDs that you order. However, you are still able to change the settings for the DID in the order page.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:DefaultDIDRouting.png|800px]]&lt;br /&gt;
&lt;br /&gt;
'''Select Plan''': Incoming Calls and Monthly Fee for the DID will be charged according to the Plan selected here. Plan prices change depending on the rate center. These prices will be shown when purchasing numbers.&lt;br /&gt;
&lt;br /&gt;
'''CallerID Name Lookup''': When activated, the system will perform a lookup in the LIBD/CNAM database to find the name matching the number of callers with a US/Canadian CallerID Number. The result of this query will be displayed with the following format &amp;quot;Caller ID name portion&amp;quot; &amp;lt;Caller ID number&amp;gt;.&lt;br /&gt;
&lt;br /&gt;
'''DID POP''': Here you can set the server in which your software/device is registered to or receiving the call from.&lt;br /&gt;
 '''Note''': Always make sure to register your software/device to the same server you set in this option to properly receive incoming calls.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:DefaultRouting.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Routing''': Here you can select the destination of the incoming calls to your DID number. You can routed directly to one account/sub account, an [[Digital Receptionist (IVR)]], [[Calling Queues]], [[Time Conditions]], [[Ring Groups]], etc.&lt;br /&gt;
&lt;br /&gt;
'''Failover Options''': This setting let you define where to redirect the call when the destination is '''Busy''', '''Unreachable''' or '''No Answer'''. These options can be displayed by clicking the '''Show Failover Options''' button&lt;br /&gt;
&lt;br /&gt;
 Note: You can change these settings later or enable additional settings in your Customer Portal&amp;gt;&amp;gt;DID Numbers&amp;gt;&amp;gt;Manage DID(s)&lt;br /&gt;
&lt;br /&gt;
== Newsletter ==&lt;br /&gt;
&lt;br /&gt;
'''Newsletter Subscription''': You can set if you want to receive news from voip.ms&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accset news.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
== Advanced ==&lt;br /&gt;
&lt;br /&gt;
These options allow you to set the allowed codecs, DTMF modes and NAT options. These settings are recommended for advanced users only.&lt;br /&gt;
&lt;br /&gt;
[[File:Accset adv.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''NAT (Network Address Translation)''': This setting should be set to Yes if you're behind a NAT, if not set to No. If you're unsure what setting means, is highly recommended that you leave it to Yes.&lt;br /&gt;
&lt;br /&gt;
'''DTMF Mode''': This allows you to select the DTMF mode that is going to be used for your account. If you set this to AUTO the RFC2833 (AVT) is going to be used and automatically switch to INBAND if the other end doesn't support RFC2833.&lt;br /&gt;
 Note: Its recommended that you select the same DTMF mode in your device. &lt;br /&gt;
&lt;br /&gt;
'''Allowed Codecs''': This setting allows you to select the codecs that you're going to use, you can choose to allow one or more codecs at the time.&lt;br /&gt;
 Note: It's recommended that you select Allow All and only change it if you have an specific reason to do so.&lt;br /&gt;
&lt;br /&gt;
[[category:guides]]&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/Account_Settings</id>
		<title>Account Settings</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/Account_Settings"/>
				<updated>2017-02-14T21:03:27Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;The Account Settings page is one of the most important pages in the Customer Portal, it will help you out as its name points out, to set all the proper ways to use your account, and these settings will affect the global settings on your account, from Device type to routing and even security of your account. &lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
__TOC__&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
== Account Routing ==&lt;br /&gt;
&lt;br /&gt;
These settings define the routes the system will use when you place calls to either US, Canada, Toll Free or International Numbers.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accsettings1.png]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Canada Routing''': You can choose between the [[Value vs Premium|'''Value''' or '''Premium''']] Route. If you are looking for the best wholesale price without any volume commitment you can use the '''Value''' route, however the CallerID number is not always guaranteed to work in this route.&lt;br /&gt;
The '''Premium''' route has a better level of quality, at a price that is a little higher than the value option. The CallerID Number is guaranteed to work for this route. Please note that the '''Canada Route''' can also be set independently per sub account, under the [[Sub Accounts]] settings. &lt;br /&gt;
&lt;br /&gt;
  Please note the US route is only available on the Premium route with a rate of $0.01 (one cent) per minute.&lt;br /&gt;
&lt;br /&gt;
'''International Routing''': You can choose also the [[Value vs Premium|'''Value''' or '''Premium''']] route. Both routes are reliable, however you're going to find better quality if you're using the '''Premium''' route. The CallerID Number is not guaranteed to work for international calls even if you use the '''Premium''' route.&lt;br /&gt;
&lt;br /&gt;
'''Toll Free Routing''': This define the route the system will use when you place a Toll-Free call to Canada or USA. You can choose between the '''Value''' route (free to use) or the '''Premium''' (at 1.06 cent per minute). As the other settings, the '''Premium''' route give you better quality and the CallerID Number is guaranteed to work.&lt;br /&gt;
&lt;br /&gt;
'''Toll Free Carrier''': This setting defines the type of carrier that will be used when you dial a Toll-Free number. You can choose between '''Server Default''', '''American Carrier''' or '''Canadian Carrier'''. &lt;br /&gt;
&lt;br /&gt;
 '''Server Default:''' The Carrier for the termination of your toll-free call will be chosen depending on the geographical location of the server &lt;br /&gt;
   you are registering or connecting to. American Servers will dial through an American Carrier and Canadian servers will use a Canadian Toll-Free carrier.&lt;br /&gt;
 '''American Carrier:''' Your Toll-Free calls will always go through an American Carrier, even if you use a Canadian server. &lt;br /&gt;
 '''Canadian Carrier:''' Your Toll-Free calls will always go through a Canadian Carrier, even if you use an American server.&lt;br /&gt;
&lt;br /&gt;
For example, this feature can be useful for west based Canadian customers who wish to use the Seattle server but have their toll-free termination calls go through a Canadian carrier.&lt;br /&gt;
&lt;br /&gt;
 '''Note''': Please refer to [[Service Cost]] for information about the rates of the outgoing calls.&lt;br /&gt;
&lt;br /&gt;
== Account Restrictions ==&lt;br /&gt;
&lt;br /&gt;
These settings define the restrictions the system will use when you place calls to either USA, Canada or International Numbers.&lt;br /&gt;
&lt;br /&gt;
[[File:Account Restrictions.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Allow 411 dialing''': When enabled, you can call 411 directory assistance at the cost of $0.99 per call. &lt;br /&gt;
&lt;br /&gt;
 '''Note''': XXX-555-1212 is disabled within VoIP.ms. The only way to reach directory assistance is by dialing 411.&lt;br /&gt;
&lt;br /&gt;
'''Allow International Calls''': When Disabled, calls placed to destinations outside USA/Canada will be automatically rejected. If you are using [[Sub Accounts]] you need to setup this directly in the sub account.&lt;br /&gt;
&lt;br /&gt;
 '''Note''': This option is set to &amp;quot;No&amp;quot; by default for New accounts.&lt;br /&gt;
&lt;br /&gt;
'''Max. Call Time for US48/Canadian Calls''': With these setting you can choose how long the call will last when you call to US or Canada Numbers. If a call to a US or Canada Number exceeds the maximum time you have set in this option, it will be automatically hung up.&lt;br /&gt;
&lt;br /&gt;
'''Max. Call Time for International Calls''': Works exactly as the '''Max. Call Time for US48/Canadian Calls''' setting. If a call to an International Number exceeds the maximum time set, it will be automatically hung up.&lt;br /&gt;
&lt;br /&gt;
'''International Amount Restriction''': With this option you can set the maximum amount per minute for calls to international destinations. If the Per minute rate for an International destination exceeds the value you have set in this option, the call will not be connected.&lt;br /&gt;
&lt;br /&gt;
For example, let say you can set this option to $0.150 and you want to call a UK mobile number with area code 4470, the call will not go through because the per minute rate for this area code is $0.3048.&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' This settings depends on the International Routing you have choose.&lt;br /&gt;
&lt;br /&gt;
'''Allow Calls to Countries''': Here you can select Regions or Specific Countries to allow outgoing calls from your account. &lt;br /&gt;
&lt;br /&gt;
In order to allow calls to a specific country you can click the name of the region to expand the countries within it and select the desired countries only. You can select '''Select All''' so all the countries within the continent will be selected. You will also be able to search for a specific country using the '''Search Country''' field. If you call to a country not allowed in this section, the call will not be connected.&lt;br /&gt;
&lt;br /&gt;
Please make sure to only allow those destinations that you are planning to dial and avoid having unplanned destinations allowed.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:AllowCalls.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
 '''Note''': If you have international calls disabled, it will prevent calling to the international countries you have enabled in this list.&lt;br /&gt;
&lt;br /&gt;
== General ==&lt;br /&gt;
&lt;br /&gt;
These are the general settings used by the system when you make or receive calls.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:AcctSet Gen.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
'''e911 Default CallerID''': This enables you to use any CallerID you want to with your accounts but when you call E911 the number you specify here is used as your CallerID and the call is placed and completed successfully.&lt;br /&gt;
&lt;br /&gt;
'''Dialing Mode''': This setting allows you to set the way you're going to dial other numbers.&lt;br /&gt;
&lt;br /&gt;
*North America (Recommended): You can dial to countries part of the North American Numbering Plan Administration by dialing 10 or 11 digits. (with or without the 1 prefix).  You'll need to use the  00 or 011 prefix to call international numbers.&lt;br /&gt;
&lt;br /&gt;
*E164: You dial numbers by using Country Code first. You don't need to send the 00 or 011 prefix when using this mode, but it's still supported. &lt;br /&gt;
&lt;br /&gt;
'''CallerID Number''':  You can set the [[Caller ID|callerID Number]] you want to pass if you are using an [[Devices|ATA]], IP Phone or [[Softphones|Softphone]]. It is important to pass a valid [[Caller ID|caller id]] to ensure proper termination. If you have a a device capable of passing its own [[Caller ID|CallerID number]] such as a soft switch or [[PBXs|PBX]], you can leave this blank to set the [[Caller ID|CallerID Number]] on your side.&lt;br /&gt;
&lt;br /&gt;
'''Voicemail Associated to the Main Account''': You can select the mailbox associated to the main account. &lt;br /&gt;
&lt;br /&gt;
*Accessing Voice Mail: When you dial *97 from your main account, it will not prompt for the [[voicemail]] ID. It will also not prompt for the password if you have selected &amp;quot;Skip Password&amp;quot; option in the [[voicemail]] configuration. &lt;br /&gt;
&lt;br /&gt;
*Message Waiting Indicator: When there are new messages, notice of new message(s) will be sent. This will lead to different results depending on your type of adapter, soft phone or IP phone. For example, when using a  Linksys [[Devices|ATA]] adapter, it's usually a stutter tone with periodic ring, IP phones usually use a stutter dial tone when you pick up the line and are equipped with a blinking light, soft phones usually show up a Voicemail Icon in the dial display.&lt;br /&gt;
&lt;br /&gt;
 '''Note:'''This setting only affects the device registered with the main account. &lt;br /&gt;
      If you're using a [[Sub Accounts|sub account]] you need to set the '''Internal Extension Voicemail''' for that subaccount.&lt;br /&gt;
&lt;br /&gt;
'''Music On Hold''': Most IP Phones and [[Softphones|Softphones]] inform the VoIP server they are connected to when the HOLD Button is pressed. If you would like to present music to the person you place on hold, you can select it here.&amp;lt;br&amp;gt;&lt;br /&gt;
The following options for music are available:&amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
*'''No Music, Intermittent Bleep:''' Silent, but with a subtle intermittent bleep sound to let your callers know that they are still on the line while they hold.&lt;br /&gt;
*'''Away in the Tropics:''' From Hawaii to the Caribbean, these tracks deliver sounds of ukulele, steel drums, and steel guitars. &lt;br /&gt;
*'''Coffee and Sunrise:''' Uplifting without being perky, and positive without being too smiley. &lt;br /&gt;
*'''Coffee Shop Acoustic:''' Soothing, acoustic guitar tracks makes for a relaxed feel and sets a great atmosphere. &lt;br /&gt;
*'''Easy Listening:''' Smooth, casual tunes. &lt;br /&gt;
*'''Guitar Alchemy:''' Clever harmonics and progressive chord sequences to create a joyful and warming musical experience. &lt;br /&gt;
*'''Happy Endings:''' Uplifting, commercial style. Guitar, drums, ukulele, harmonica and bells. &lt;br /&gt;
*'''Light and Casual:''' Soothing and peaceful songs with a light, positive feeling. &lt;br /&gt;
*'''Orchestral Moods:''' Emotional and dramatic tales spun by violins, pianos and full orchestras. &lt;br /&gt;
*'''Piano Mix:'''  Smooth Piano &lt;br /&gt;
*'''Rock Me Easy:''' Feel-good music to create a relaxing atmosphere. &lt;br /&gt;
*'''Spa Sounds:''' Soft, slow and serene instrumentals. &lt;br /&gt;
&lt;br /&gt;
You can test the categories by dialing the following codes: &amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
'''*** 89''' Away in the Tropics &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 90''' Coffee and Sunrise &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 91''' Coffee Shop Acoustic &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 92''' Easy Listening &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 93''' Guitar Alchemy &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 94''' Happy Endings &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 95''' Light and Casual &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 96''' Orchestral Moods &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 97''' Piano Mix &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 98''' Rock Me Easy &amp;lt;br&amp;gt;&lt;br /&gt;
'''*** 99''' Spa Sounds&amp;lt;br&amp;gt;&lt;br /&gt;
'''***100''' No Music, Intermittent Bleep&amp;lt;br&amp;gt;&lt;br /&gt;
&lt;br /&gt;
== Security ==&lt;br /&gt;
&lt;br /&gt;
Here you can change the password to access your Customer Portal as well your SIP/IAX passwords for your main account. You can also change the way SIP/IAX passwords are displayed on various pages of the &amp;quot;Customer Portal&amp;quot;.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accset sec.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Display SIP and IAX password(s) in Customer Portal''': If you enable this, the SIP/IAX passwords are going to be displayed in the Customer Portal. It is recommended that you leave this disabled. Enable this if only need if you forgot a SIP/IAX password for your account or one of your subaccounts. &lt;br /&gt;
 Note: You must enter your current customer portal password in order to enable this option.&lt;br /&gt;
&lt;br /&gt;
'''Customer Portal Password''': Here you can change the password used to log in your VoIP.ms Customer Portal.&lt;br /&gt;
&lt;br /&gt;
'''Main SIP/IAX Password''': Here is where you can change the password used to register your Main Account to one of the VoIP.ms servers.&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' For default the Main SIP/IAX Password is the same you use to login in your Customer Portal.&lt;br /&gt;
&lt;br /&gt;
== Inbound Settings==&lt;br /&gt;
&lt;br /&gt;
In this tab, you can set the protocol used for inbound calls and the device you're using for your main account.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Inbound.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Protocol for inbound DIDs''': Here you can select either SIP or IAX2 protocol. This will depend on the device you want to use with your Main account. SIP is the recommended protocol.&lt;br /&gt;
&lt;br /&gt;
'''Device type''': Make sure to select the correct type of device you're going to use with the main account to properly receive incoming calls in your device.&lt;br /&gt;
&lt;br /&gt;
== Notifications ==&lt;br /&gt;
&lt;br /&gt;
In this tab you can set whether to allow or not notifications in your mail when the balance in your account reach certain amount.&lt;br /&gt;
&lt;br /&gt;
[[File:Accset not.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
'''Balance Threshold''': Here you can set an amount and when your balance goes below this amount you will receive an email to the address set in the option below. You can select a value between $1 up to $200, or if you like you can disable the notifications (this is not recommended).&lt;br /&gt;
&lt;br /&gt;
 '''Note''': Set this according to the monthly use you have in your account. &lt;br /&gt;
&lt;br /&gt;
'''Email''': Enter the email address in which you want to receive the notifications. Make sure to enter a proper email address. If you want to receive notifications in more than one email address, you can list the emails separated with commas (,).&lt;br /&gt;
&lt;br /&gt;
 '''Note''': While the balance is below your threshold, you're going to keep receiving emails until you add new funds in your account or you change the balance threshold.&lt;br /&gt;
&lt;br /&gt;
== Default DID Routing ==&lt;br /&gt;
&lt;br /&gt;
These settings let you define the preconfigured Routing options that are going to be applied to each new DIDs that you order. However, you are still able to change the settings for the DID in the order page.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:DefaultDIDRouting.png|800px]]&lt;br /&gt;
&lt;br /&gt;
'''Select Plan''': Incoming Calls and Monthly Fee for the DID will be charged according to the Plan selected here. Plan prices change depending on the rate center. These prices will be shown when purchasing numbers.&lt;br /&gt;
&lt;br /&gt;
'''CallerID Name Lookup''': When activated, the system will perform a lookup in the LIBD/CNAM database to find the name matching the number of callers with a US/Canadian CallerID Number. The result of this query will be displayed with the following format &amp;quot;Caller ID name portion&amp;quot; &amp;lt;Caller ID number&amp;gt;.&lt;br /&gt;
&lt;br /&gt;
'''DID POP''': Here you can set the server in which your software/device is registered to or receiving the call from.&lt;br /&gt;
 '''Note''': Always make sure to register your software/device to the same server you set in this option to properly receive incoming calls.&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:DefaultRouting.png|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''Routing''': Here you can select the destination of the incoming calls to your DID number. You can routed directly to one account/sub account, an [[Digital Receptionist (IVR)]], [[Calling Queues]], [[Time Conditions]], [[Ring Groups]], etc.&lt;br /&gt;
&lt;br /&gt;
'''Failover Options''': This setting let you define where to redirect the call when the destination is '''Busy''', '''Unreachable''' or '''No Answer'''. These options can be displayed by clicking the '''Show Failover Options''' button&lt;br /&gt;
&lt;br /&gt;
 Note: You can change these settings later or enable additional settings in your Customer Portal&amp;gt;&amp;gt;DID Numbers&amp;gt;&amp;gt;Manage DID(s)&lt;br /&gt;
&lt;br /&gt;
== Newsletter ==&lt;br /&gt;
&lt;br /&gt;
'''Newsletter Subscription''': You can set if you want to receive news from voip.ms&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[File:Accset news.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
== Advanced ==&lt;br /&gt;
&lt;br /&gt;
These options allow you to set the allowed codecs, DTMF modes and NAT options. These settings are recommended for advanced users only.&lt;br /&gt;
&lt;br /&gt;
[[File:Accset adv.jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''NAT (Network Address Translation)''': This setting should be set to Yes if you're behind a NAT, if not set to No. If you're unsure what setting means, is highly recommended that you leave it to Yes.&lt;br /&gt;
&lt;br /&gt;
'''DTMF Mode''': This allows you to select the DTMF mode that is going to be used for your account. If you set this to AUTO the RFC2833 (AVT) is going to be used and automatically switch to INBAND if the other end doesn't support RFC2833.&lt;br /&gt;
 Note: Its recommended that you select the same DTMF mode in your device. &lt;br /&gt;
&lt;br /&gt;
'''Allowed Codecs''': This setting allows you to select the codecs that you're going to use, you can choose to allow one or more codecs at the time.&lt;br /&gt;
 Note: It's recommended that you select Allow All and only change it if you have an specific reason to do so.&lt;br /&gt;
&lt;br /&gt;
[[category:guides]]&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/File:Accsettings1.png</id>
		<title>File:Accsettings1.png</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/File:Accsettings1.png"/>
				<updated>2017-02-14T21:03:06Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
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		<author><name>William</name></author>	</entry>

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		<id>https://wiki.voip.ms/article/File:SonicWall5.png</id>
		<title>File:SonicWall5.png</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/File:SonicWall5.png"/>
				<updated>2014-10-13T14:55:33Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
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		<author><name>William</name></author>	</entry>

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		<id>https://wiki.voip.ms/article/File:SonicWall4.png</id>
		<title>File:SonicWall4.png</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/File:SonicWall4.png"/>
				<updated>2014-10-13T14:55:18Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
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		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/File:SonicWall3.png</id>
		<title>File:SonicWall3.png</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/File:SonicWall3.png"/>
				<updated>2014-10-13T14:54:52Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
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		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/File:SonicWall2.png</id>
		<title>File:SonicWall2.png</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/File:SonicWall2.png"/>
				<updated>2014-10-13T14:54:40Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/SonicWall</id>
		<title>SonicWall</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/SonicWall"/>
				<updated>2014-10-13T14:53:40Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;Author: James A. Russo jr@halo3.net / Halo3 Consulting, LLC&lt;br /&gt;
&lt;br /&gt;
'''''Synopsis:''''' &lt;br /&gt;
&lt;br /&gt;
	When using a SonicWALL and a PBX behind that SonicWALL, some of the inbound SIP connections may get refused because the SonicWALL is quick to timeout the UDP sessions on the firewall. This will result in a situation where some incoming calls connect just fine, but then others just a minute or so later would timeout and never connect.&lt;br /&gt;
&lt;br /&gt;
In our configuration we are using a TZ-210 running SonicOS Enhanced 5.8.1.13-1o. However, the same configuration can likely be done on various SonicWALL devices. &lt;br /&gt;
&lt;br /&gt;
We were able to determine what was happening by watching the logs on the Sonicwall where would find dropped UDP packets originating from the VOIP.MS server on port 5060 to our WAN ip address on some various UDP port . &lt;br /&gt;
&lt;br /&gt;
'''''Solution:'''''&lt;br /&gt;
&lt;br /&gt;
The solution will be to add a firewall rule from LAN-&amp;gt;WAN which will apply to the Internal LAN PBX IP to the Address group of the VOIP.MS servers. This will be an Allow Firewall rule, but more importantly will define the UDP session timeout to be 500 seconds (vs the normal 30 seconds).&lt;br /&gt;
&lt;br /&gt;
'''''Step 1:  Creating the Address Objects'''''&lt;br /&gt;
&lt;br /&gt;
Create the address object for all the various VOIP.MS servers you may connect to. You should list your primary servers and any secondary servers you may connect to. You don’t want to fail over to a secondary server and then have to remember to modify your firewall rules. &lt;br /&gt;
&lt;br /&gt;
These will be on the WAN zone, and should be FQDN objects. We do this so that if the IP address of the voip.ms server ever should change the rule will still work. &lt;br /&gt;
&lt;br /&gt;
Repeat this for any other VOIP.MS servers you '''''may''''' connect to.&lt;br /&gt;
&lt;br /&gt;
[[File:SonicWall.png|1000px|thumb|left]]&lt;br /&gt;
&lt;br /&gt;
'''''Step 2: Create the Address Group Objects'''''&lt;br /&gt;
&lt;br /&gt;
Create an Address Group Object that will contain all of the addresses you defined in Step 1.  This will be the actual object we will use in the firewall rule. &lt;br /&gt;
&lt;br /&gt;
[[File:SonicWall2.png|1000px|thumb|left]]&lt;br /&gt;
&lt;br /&gt;
'''''Step 3: Create Address Object for the PBX which is behind the SonicWALL.'''''&lt;br /&gt;
&lt;br /&gt;
This step is technically optional, as in the firewall rule you could always just apply this firewall rule from ANY host in the network to the VOIP.MS servers. By including this rule, the UDP timeout will only be extended for sessions created from the PBX to the VOIP.MS servers.  I am not convinced that this is really necessary or enhances security that much. In our configuration we have one PBX internally behind the SonicWALL. If you have many phones behind the SonicWALL, you may want to just skip this step and specify ANY as the source address in Step 4 below. &lt;br /&gt;
&lt;br /&gt;
[[File:SonicWall3.png|1000px|thumb|left]]&lt;br /&gt;
&lt;br /&gt;
'''''Step 4: Create the Firewall Rule'''''&lt;br /&gt;
&lt;br /&gt;
In this step you will create the firewall rule that will allow access from LAN -&amp;gt; WAN and also adjust the specific UDP timeout. This is a redundant rule as there is already a firewall rule that permits LAN -&amp;gt; WAN. However, that rule will include the 30-second UDP timeout that is the actual cause of the problem.&lt;br /&gt;
&lt;br /&gt;
[[File:SonicWall4.png|1000px|thumb|left]]&lt;br /&gt;
&lt;br /&gt;
Once you have setup the rule, click on the Advanced tab and adjust the UDP timeout value to something like 300 seconds ( 5 minutes). This should prevent the SonicWALL from dropping the INVITE sip packets which arrives more than 30 seconds since the last outgoing SIP packet to the VOIP.MS server. &lt;br /&gt;
&lt;br /&gt;
[[File:SonicWall5.png|1000px|thumb|left]]&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/SonicWall</id>
		<title>SonicWall</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/SonicWall"/>
				<updated>2014-10-13T14:51:26Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;Author: James A. Russo jr@halo3.net / Halo3 Consulting, LLC&lt;br /&gt;
&lt;br /&gt;
'''''Synopsis:''''' &lt;br /&gt;
&lt;br /&gt;
	When using a SonicWALL and a PBX behind that SonicWALL, some of the inbound SIP connections may get refused because the SonicWALL is quick to timeout the UDP sessions on the firewall. This will result in a situation where some incoming calls connect just fine, but then others just a minute or so later would timeout and never connect.&lt;br /&gt;
&lt;br /&gt;
In our configuration we are using a TZ-210 running SonicOS Enhanced 5.8.1.13-1o. However, the same configuration can likely be done on various SonicWALL devices. &lt;br /&gt;
&lt;br /&gt;
We were able to determine what was happening by watching the logs on the Sonicwall where would find dropped UDP packets originating from the VOIP.MS server on port 5060 to our WAN ip address on some various UDP port . &lt;br /&gt;
&lt;br /&gt;
'''''Solution:'''''&lt;br /&gt;
&lt;br /&gt;
The solution will be to add a firewall rule from LAN-&amp;gt;WAN which will apply to the Internal LAN PBX IP to the Address group of the VOIP.MS servers. This will be an Allow Firewall rule, but more importantly will define the UDP session timeout to be 500 seconds (vs the normal 30 seconds).&lt;br /&gt;
&lt;br /&gt;
'''''Step 1:  Creating the Address Objects'''''&lt;br /&gt;
&lt;br /&gt;
Create the address object for all the various VOIP.MS servers you may connect to. You should list your primary servers and any secondary servers you may connect to. You don’t want to fail over to a secondary server and then have to remember to modify your firewall rules. &lt;br /&gt;
&lt;br /&gt;
These will be on the WAN zone, and should be FQDN objects. We do this so that if the IP address of the voip.ms server ever should change the rule will still work. &lt;br /&gt;
&lt;br /&gt;
Repeat this for any other VOIP.MS servers you '''''may''''' connect to.&lt;br /&gt;
&lt;br /&gt;
[[File:SonicWall.png|1000px|thumb|left]]&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/SonicWall</id>
		<title>SonicWall</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/SonicWall"/>
				<updated>2014-10-13T14:51:15Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;Author: James A. Russo jr@halo3.net / Halo3 Consulting, LLC&lt;br /&gt;
&lt;br /&gt;
'''''Synopsis:''''' &lt;br /&gt;
&lt;br /&gt;
	When using a SonicWALL and a PBX behind that SonicWALL, some of the inbound SIP connections may get refused because the SonicWALL is quick to timeout the UDP sessions on the firewall. This will result in a situation where some incoming calls connect just fine, but then others just a minute or so later would timeout and never connect.&lt;br /&gt;
&lt;br /&gt;
In our configuration we are using a TZ-210 running SonicOS Enhanced 5.8.1.13-1o. However, the same configuration can likely be done on various SonicWALL devices. &lt;br /&gt;
&lt;br /&gt;
We were able to determine what was happening by watching the logs on the Sonicwall where would find dropped UDP packets originating from the VOIP.MS server on port 5060 to our WAN ip address on some various UDP port . &lt;br /&gt;
&lt;br /&gt;
'''''Solution:'''''&lt;br /&gt;
&lt;br /&gt;
The solution will be to add a firewall rule from LAN-&amp;gt;WAN which will apply to the Internal LAN PBX IP to the Address group of the VOIP.MS servers. This will be an Allow Firewall rule, but more importantly will define the UDP session timeout to be 500 seconds (vs the normal 30 seconds).&lt;br /&gt;
&lt;br /&gt;
'''''Step 1:  Creating the Address Objects'''''&lt;br /&gt;
&lt;br /&gt;
Create the address object for all the various VOIP.MS servers you may connect to. You should list your primary servers and any secondary servers you may connect to. You don’t want to fail over to a secondary server and then have to remember to modify your firewall rules. &lt;br /&gt;
&lt;br /&gt;
These will be on the WAN zone, and should be FQDN objects. We do this so that if the IP address of the voip.ms server ever should change the rule will still work. &lt;br /&gt;
&lt;br /&gt;
Repeat this for any other VOIP.MS servers you '''''may''''' connect to.&lt;br /&gt;
&lt;br /&gt;
[[File:SonicWall.png|1100px|thumb|left]]&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/SonicWall</id>
		<title>SonicWall</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/SonicWall"/>
				<updated>2014-10-13T14:51:02Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;Author: James A. Russo jr@halo3.net / Halo3 Consulting, LLC&lt;br /&gt;
&lt;br /&gt;
'''''Synopsis:''''' &lt;br /&gt;
&lt;br /&gt;
	When using a SonicWALL and a PBX behind that SonicWALL, some of the inbound SIP connections may get refused because the SonicWALL is quick to timeout the UDP sessions on the firewall. This will result in a situation where some incoming calls connect just fine, but then others just a minute or so later would timeout and never connect.&lt;br /&gt;
&lt;br /&gt;
In our configuration we are using a TZ-210 running SonicOS Enhanced 5.8.1.13-1o. However, the same configuration can likely be done on various SonicWALL devices. &lt;br /&gt;
&lt;br /&gt;
We were able to determine what was happening by watching the logs on the Sonicwall where would find dropped UDP packets originating from the VOIP.MS server on port 5060 to our WAN ip address on some various UDP port . &lt;br /&gt;
&lt;br /&gt;
'''''Solution:'''''&lt;br /&gt;
&lt;br /&gt;
The solution will be to add a firewall rule from LAN-&amp;gt;WAN which will apply to the Internal LAN PBX IP to the Address group of the VOIP.MS servers. This will be an Allow Firewall rule, but more importantly will define the UDP session timeout to be 500 seconds (vs the normal 30 seconds).&lt;br /&gt;
&lt;br /&gt;
'''''Step 1:  Creating the Address Objects'''''&lt;br /&gt;
&lt;br /&gt;
Create the address object for all the various VOIP.MS servers you may connect to. You should list your primary servers and any secondary servers you may connect to. You don’t want to fail over to a secondary server and then have to remember to modify your firewall rules. &lt;br /&gt;
&lt;br /&gt;
These will be on the WAN zone, and should be FQDN objects. We do this so that if the IP address of the voip.ms server ever should change the rule will still work. &lt;br /&gt;
&lt;br /&gt;
Repeat this for any other VOIP.MS servers you '''''may''''' connect to.&lt;br /&gt;
&lt;br /&gt;
[[File:SonicWall.png|1200px|thumb|left]]&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/SonicWall</id>
		<title>SonicWall</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/SonicWall"/>
				<updated>2014-10-13T14:50:47Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;Author: James A. Russo jr@halo3.net / Halo3 Consulting, LLC&lt;br /&gt;
&lt;br /&gt;
'''''Synopsis:''''' &lt;br /&gt;
&lt;br /&gt;
	When using a SonicWALL and a PBX behind that SonicWALL, some of the inbound SIP connections may get refused because the SonicWALL is quick to timeout the UDP sessions on the firewall. This will result in a situation where some incoming calls connect just fine, but then others just a minute or so later would timeout and never connect.&lt;br /&gt;
&lt;br /&gt;
In our configuration we are using a TZ-210 running SonicOS Enhanced 5.8.1.13-1o. However, the same configuration can likely be done on various SonicWALL devices. &lt;br /&gt;
&lt;br /&gt;
We were able to determine what was happening by watching the logs on the Sonicwall where would find dropped UDP packets originating from the VOIP.MS server on port 5060 to our WAN ip address on some various UDP port . &lt;br /&gt;
&lt;br /&gt;
'''''Solution:'''''&lt;br /&gt;
&lt;br /&gt;
The solution will be to add a firewall rule from LAN-&amp;gt;WAN which will apply to the Internal LAN PBX IP to the Address group of the VOIP.MS servers. This will be an Allow Firewall rule, but more importantly will define the UDP session timeout to be 500 seconds (vs the normal 30 seconds).&lt;br /&gt;
&lt;br /&gt;
'''''Step 1:  Creating the Address Objects'''''&lt;br /&gt;
&lt;br /&gt;
Create the address object for all the various VOIP.MS servers you may connect to. You should list your primary servers and any secondary servers you may connect to. You don’t want to fail over to a secondary server and then have to remember to modify your firewall rules. &lt;br /&gt;
&lt;br /&gt;
These will be on the WAN zone, and should be FQDN objects. We do this so that if the IP address of the voip.ms server ever should change the rule will still work. &lt;br /&gt;
&lt;br /&gt;
Repeat this for any other VOIP.MS servers you '''''may''''' connect to.&lt;br /&gt;
&lt;br /&gt;
[[File:SonicWall.png|900px|thumb|left]]&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/SonicWall</id>
		<title>SonicWall</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/SonicWall"/>
				<updated>2014-10-13T14:50:27Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;Author: James A. Russo jr@halo3.net / Halo3 Consulting, LLC&lt;br /&gt;
&lt;br /&gt;
'''''Synopsis:''''' &lt;br /&gt;
&lt;br /&gt;
	When using a SonicWALL and a PBX behind that SonicWALL, some of the inbound SIP connections may get refused because the SonicWALL is quick to timeout the UDP sessions on the firewall. This will result in a situation where some incoming calls connect just fine, but then others just a minute or so later would timeout and never connect.&lt;br /&gt;
&lt;br /&gt;
In our configuration we are using a TZ-210 running SonicOS Enhanced 5.8.1.13-1o. However, the same configuration can likely be done on various SonicWALL devices. &lt;br /&gt;
&lt;br /&gt;
We were able to determine what was happening by watching the logs on the Sonicwall where would find dropped UDP packets originating from the VOIP.MS server on port 5060 to our WAN ip address on some various UDP port . &lt;br /&gt;
&lt;br /&gt;
'''''Solution:'''''&lt;br /&gt;
&lt;br /&gt;
The solution will be to add a firewall rule from LAN-&amp;gt;WAN which will apply to the Internal LAN PBX IP to the Address group of the VOIP.MS servers. This will be an Allow Firewall rule, but more importantly will define the UDP session timeout to be 500 seconds (vs the normal 30 seconds).&lt;br /&gt;
&lt;br /&gt;
'''''Step 1:  Creating the Address Objects'''''&lt;br /&gt;
&lt;br /&gt;
Create the address object for all the various VOIP.MS servers you may connect to. You should list your primary servers and any secondary servers you may connect to. You don’t want to fail over to a secondary server and then have to remember to modify your firewall rules. &lt;br /&gt;
&lt;br /&gt;
These will be on the WAN zone, and should be FQDN objects. We do this so that if the IP address of the voip.ms server ever should change the rule will still work. &lt;br /&gt;
&lt;br /&gt;
Repeat this for any other VOIP.MS servers you '''''may''''' connect to.&lt;br /&gt;
&lt;br /&gt;
[[File:SonicWall.png|700px|thumb|left]]&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/SonicWall</id>
		<title>SonicWall</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/SonicWall"/>
				<updated>2014-10-13T14:49:53Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;Author: James A. Russo jr@halo3.net / Halo3 Consulting, LLC&lt;br /&gt;
&lt;br /&gt;
'''''Synopsis:''''' &lt;br /&gt;
&lt;br /&gt;
	When using a SonicWALL and a PBX behind that SonicWALL, some of the inbound SIP connections may get refused because the SonicWALL is quick to timeout the UDP sessions on the firewall. This will result in a situation where some incoming calls connect just fine, but then others just a minute or so later would timeout and never connect.&lt;br /&gt;
&lt;br /&gt;
In our configuration we are using a TZ-210 running SonicOS Enhanced 5.8.1.13-1o. However, the same configuration can likely be done on various SonicWALL devices. &lt;br /&gt;
&lt;br /&gt;
We were able to determine what was happening by watching the logs on the Sonicwall where would find dropped UDP packets originating from the VOIP.MS server on port 5060 to our WAN ip address on some various UDP port . &lt;br /&gt;
&lt;br /&gt;
'''''Solution:'''''&lt;br /&gt;
&lt;br /&gt;
The solution will be to add a firewall rule from LAN-&amp;gt;WAN which will apply to the Internal LAN PBX IP to the Address group of the VOIP.MS servers. This will be an Allow Firewall rule, but more importantly will define the UDP session timeout to be 500 seconds (vs the normal 30 seconds).&lt;br /&gt;
&lt;br /&gt;
'''''Step 1:  Creating the Address Objects'''''&lt;br /&gt;
&lt;br /&gt;
Create the address object for all the various VOIP.MS servers you may connect to. You should list your primary servers and any secondary servers you may connect to. You don’t want to fail over to a secondary server and then have to remember to modify your firewall rules. &lt;br /&gt;
&lt;br /&gt;
These will be on the WAN zone, and should be FQDN objects. We do this so that if the IP address of the voip.ms server ever should change the rule will still work. &lt;br /&gt;
&lt;br /&gt;
Repeat this for any other VOIP.MS servers you '''''may''''' connect to.&lt;br /&gt;
&lt;br /&gt;
[[File:SonicWall.png|200px|thumb|left|alt text]]&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/SonicWall</id>
		<title>SonicWall</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/SonicWall"/>
				<updated>2014-10-13T14:49:23Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;Author: James A. Russo jr@halo3.net / Halo3 Consulting, LLC&lt;br /&gt;
&lt;br /&gt;
'''''Synopsis:''''' &lt;br /&gt;
&lt;br /&gt;
	When using a SonicWALL and a PBX behind that SonicWALL, some of the inbound SIP connections may get refused because the SonicWALL is quick to timeout the UDP sessions on the firewall. This will result in a situation where some incoming calls connect just fine, but then others just a minute or so later would timeout and never connect.&lt;br /&gt;
&lt;br /&gt;
In our configuration we are using a TZ-210 running SonicOS Enhanced 5.8.1.13-1o. However, the same configuration can likely be done on various SonicWALL devices. &lt;br /&gt;
&lt;br /&gt;
We were able to determine what was happening by watching the logs on the Sonicwall where would find dropped UDP packets originating from the VOIP.MS server on port 5060 to our WAN ip address on some various UDP port . &lt;br /&gt;
&lt;br /&gt;
'''''Solution:'''''&lt;br /&gt;
&lt;br /&gt;
The solution will be to add a firewall rule from LAN-&amp;gt;WAN which will apply to the Internal LAN PBX IP to the Address group of the VOIP.MS servers. This will be an Allow Firewall rule, but more importantly will define the UDP session timeout to be 500 seconds (vs the normal 30 seconds).&lt;br /&gt;
&lt;br /&gt;
'''''Step 1:  Creating the Address Objects'''''&lt;br /&gt;
&lt;br /&gt;
Create the address object for all the various VOIP.MS servers you may connect to. You should list your primary servers and any secondary servers you may connect to. You don’t want to fail over to a secondary server and then have to remember to modify your firewall rules. &lt;br /&gt;
&lt;br /&gt;
These will be on the WAN zone, and should be FQDN objects. We do this so that if the IP address of the voip.ms server ever should change the rule will still work. &lt;br /&gt;
&lt;br /&gt;
Repeat this for any other VOIP.MS servers you '''''may''''' connect to.&lt;br /&gt;
&lt;br /&gt;
[[File:SonicWall.png]]&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/File:SonicWall.png</id>
		<title>File:SonicWall.png</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/File:SonicWall.png"/>
				<updated>2014-10-13T14:49:02Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/SonicWall</id>
		<title>SonicWall</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/SonicWall"/>
				<updated>2014-10-13T14:48:00Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;Author: James A. Russo jr@halo3.net / Halo3 Consulting, LLC&lt;br /&gt;
&lt;br /&gt;
'''''Synopsis:''''' &lt;br /&gt;
&lt;br /&gt;
	When using a SonicWALL and a PBX behind that SonicWALL, some of the inbound SIP connections may get refused because the SonicWALL is quick to timeout the UDP sessions on the firewall. This will result in a situation where some incoming calls connect just fine, but then others just a minute or so later would timeout and never connect.&lt;br /&gt;
&lt;br /&gt;
In our configuration we are using a TZ-210 running SonicOS Enhanced 5.8.1.13-1o. However, the same configuration can likely be done on various SonicWALL devices. &lt;br /&gt;
&lt;br /&gt;
We were able to determine what was happening by watching the logs on the Sonicwall where would find dropped UDP packets originating from the VOIP.MS server on port 5060 to our WAN ip address on some various UDP port . &lt;br /&gt;
&lt;br /&gt;
'''''Solution:'''''&lt;br /&gt;
&lt;br /&gt;
The solution will be to add a firewall rule from LAN-&amp;gt;WAN which will apply to the Internal LAN PBX IP to the Address group of the VOIP.MS servers. This will be an Allow Firewall rule, but more importantly will define the UDP session timeout to be 500 seconds (vs the normal 30 seconds).&lt;br /&gt;
&lt;br /&gt;
'''''Step 1:  Creating the Address Objects'''''&lt;br /&gt;
&lt;br /&gt;
Create the address object for all the various VOIP.MS servers you may connect to. You should list your primary servers and any secondary servers you may connect to. You don’t want to fail over to a secondary server and then have to remember to modify your firewall rules. &lt;br /&gt;
&lt;br /&gt;
These will be on the WAN zone, and should be FQDN objects. We do this so that if the IP address of the voip.ms server ever should change the rule will still work. &lt;br /&gt;
&lt;br /&gt;
Repeat this for any other VOIP.MS servers you '''''may''''' connect to.&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/SonicWall</id>
		<title>SonicWall</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/SonicWall"/>
				<updated>2014-10-13T14:47:38Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;Author: James A. Russo jr@halo3.net / Halo3 Consulting, LLC&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
'''''Synopsis:''''' &lt;br /&gt;
&lt;br /&gt;
	When using a SonicWALL and a PBX behind that SonicWALL, some of the inbound SIP connections may get refused because the SonicWALL is quick to timeout the UDP sessions on the firewall. This will result in a situation where some incoming calls connect just fine, but then others just a minute or so later would timeout and never connect.&lt;br /&gt;
&lt;br /&gt;
In our configuration we are using a TZ-210 running SonicOS Enhanced 5.8.1.13-1o. However, the same configuration can likely be done on various SonicWALL devices. &lt;br /&gt;
&lt;br /&gt;
We were able to determine what was happening by watching the logs on the Sonicwall where would find dropped UDP packets originating from the VOIP.MS server on port 5060 to our WAN ip address on some various UDP port . &lt;br /&gt;
&lt;br /&gt;
'''''Solution:'''''&lt;br /&gt;
&lt;br /&gt;
The solution will be to add a firewall rule from LAN-&amp;gt;WAN which will apply to the Internal LAN PBX IP to the Address group of the VOIP.MS servers. This will be an Allow Firewall rule, but more importantly will define the UDP session timeout to be 500 seconds (vs the normal 30 seconds).&lt;br /&gt;
&lt;br /&gt;
'''''Step 1:  Creating the Address Objects'''''&lt;br /&gt;
&lt;br /&gt;
Create the address object for all the various VOIP.MS servers you may connect to. You should list your primary servers and any secondary servers you may connect to. You don’t want to fail over to a secondary server and then have to remember to modify your firewall rules. &lt;br /&gt;
&lt;br /&gt;
These will be on the WAN zone, and should be FQDN objects. We do this so that if the IP address of the voip.ms server ever should change the rule will still work. &lt;br /&gt;
&lt;br /&gt;
Repeat this for any other VOIP.MS servers you '''''may''''' connect to.&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/SonicWall</id>
		<title>SonicWall</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/SonicWall"/>
				<updated>2014-10-13T14:47:10Z</updated>
		
		<summary type="html">&lt;p&gt;William: Created page with &amp;quot;Author: James A. Russo jr@halo3.net / Halo3 Consulting, LLC   Synopsis:   	When using a SonicWALL and a PBX behind that SonicWALL, some of the inbound SIP connections may get ref...&amp;quot;&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;Author: James A. Russo jr@halo3.net / Halo3 Consulting, LLC&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
Synopsis: &lt;br /&gt;
&lt;br /&gt;
	When using a SonicWALL and a PBX behind that SonicWALL, some of the inbound SIP connections may get refused because the SonicWALL is quick to timeout the UDP sessions on the firewall. This will result in a situation where some incoming calls connect just fine, but then others just a minute or so later would timeout and never connect.&lt;br /&gt;
&lt;br /&gt;
In our configuration we are using a TZ-210 running SonicOS Enhanced 5.8.1.13-1o. However, the same configuration can likely be done on various SonicWALL devices. &lt;br /&gt;
&lt;br /&gt;
We were able to determine what was happening by watching the logs on the Sonicwall where would find dropped UDP packets originating from the VOIP.MS server on port 5060 to our WAN ip address on some various UDP port . &lt;br /&gt;
&lt;br /&gt;
Solution:&lt;br /&gt;
&lt;br /&gt;
The solution will be to add a firewall rule from LAN-&amp;gt;WAN which will apply to the Internal LAN PBX IP to the Address group of the VOIP.MS servers. This will be an Allow Firewall rule, but more importantly will define the UDP session timeout to be 500 seconds (vs the normal 30 seconds).&lt;br /&gt;
&lt;br /&gt;
'''''Step 1:  Creating the Address Objects'''''&lt;br /&gt;
&lt;br /&gt;
Create the address object for all the various VOIP.MS servers you may connect to. You should list your primary servers and any secondary servers you may connect to. You don’t want to fail over to a secondary server and then have to remember to modify your firewall rules. &lt;br /&gt;
&lt;br /&gt;
These will be on the WAN zone, and should be FQDN objects. We do this so that if the IP address of the voip.ms server ever should change the rule will still work. &lt;br /&gt;
&lt;br /&gt;
Repeat this for any other VOIP.MS servers you '''''may''''' connect to.&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/TalkSwitch</id>
		<title>TalkSwitch</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/TalkSwitch"/>
				<updated>2014-08-29T15:26:50Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt; '''Disclaimer:''' Talkswitch Fortivoice System usage is not 100% guaranteed to work with VoIP.ms, the following&lt;br /&gt;
 configuration is the result of several real tests and feedback from our customers, this is by far the most reliable&lt;br /&gt;
 configuration that we have been able to find out,  please follow the instructions as explained, and contact us if you&lt;br /&gt;
 have any doubts.&lt;br /&gt;
&lt;br /&gt;
==='''Configuring TalkSwitch for VoIP.ms VoIP Service'''===&lt;br /&gt;
&lt;br /&gt;
This guide will show you how to set up VoIP.ms VoIP service.&lt;br /&gt;
&lt;br /&gt;
When you open an account with VoIP.ms, we will provide you with account credentials information. Use this information to set up the service provider profile and VoIP numbers on your TalkSwitch system.&lt;br /&gt;
&lt;br /&gt;
''Setting up a service provider profile'' &lt;br /&gt;
&lt;br /&gt;
1. Select the VoIP Configuration page. &lt;br /&gt;
&lt;br /&gt;
[[File:Talkswitch1.png]]&lt;br /&gt;
&lt;br /&gt;
2. Select a Profile (SP 1 to SP 4) that you wish to assign for FortiCall.&lt;br /&gt;
&lt;br /&gt;
3. Select the Activate Profile checkbox.&lt;br /&gt;
&lt;br /&gt;
4. Under Service Provider select Manual Setup.&lt;br /&gt;
&lt;br /&gt;
5. Click the Update Configuration button. The essential settings will be completed automatically.&lt;br /&gt;
&lt;br /&gt;
Disable public IP address substitution leave this unchecked.&lt;br /&gt;
Check Register with authentication username.&lt;br /&gt;
Check Enable NAT Keep alives (You can keep the default NAT settings)&lt;br /&gt;
Codec Options: Here make sure to select only the codec supported for VoIP.ms (g.729 and g.711u). And if you have enough bandwidth you should select the g.711u as Preferred codec.&lt;br /&gt;
&lt;br /&gt;
[[File:Image002.png]]&lt;br /&gt;
&lt;br /&gt;
===Provisioning Details===&lt;br /&gt;
*'''Proxy Server name''': server.voip.ms (one of the voip.ms servers, e.g. atlanta.voip.ms)&lt;br /&gt;
*'''Registrar server name''': same as above&lt;br /&gt;
*'''Outbound proxy''': same as above&lt;br /&gt;
*'''Realm/domain''': same as above&lt;br /&gt;
&lt;br /&gt;
==Configuring Outgoing calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
*Select the '''VoIP Numbers''' page&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number outbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
*Select one of the empty voip number slots and then check the '''Activate VoIP number'''&lt;br /&gt;
*'''Select a VoIP profile''': Here select the profile to create for VoIP.ms&lt;br /&gt;
&lt;br /&gt;
'''''The record for your main account sip userID as shown in the screen, this is required as this section sends the correct information required from our servers on the INVITES:FROM which will be the userid, if you don't add this record the Talkswitch will be sending the number (DID set) instead and this will result in continuous 'deregistrations', registration dropping.'''''&lt;br /&gt;
&lt;br /&gt;
===Username and Password===&lt;br /&gt;
&lt;br /&gt;
*'''User/Account''': 100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)&lt;br /&gt;
&lt;br /&gt;
*'''Password''': (the password for your account. If you're using the main account this is the same you use to login to your Customer Portal, unless you change the password, if using a sub account you'll notice each sub account has their own password)&lt;br /&gt;
&lt;br /&gt;
 '''Note''' You can only use the main account for outgoing calls, this is due to the Talkswitch system not supporting&lt;br /&gt;
 the Underscore on the sub accounts.&lt;br /&gt;
&lt;br /&gt;
==Configuring Incoming calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' You don't need to purchase a DID number for outgoing calls. The VoIP number is used as your CallerID number.&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number inbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
Here you can set the callerID number that you want (or your DID number if you want to receive incoming calls), you need to enter the number filling in the 3 fields. &lt;br /&gt;
*'''Country Code''': 1 &lt;br /&gt;
*'''City or area code''': 555&lt;br /&gt;
*'''Number''': 4443322&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[Category:PBXes]]&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' If you are planning to use more than 1 number with us, as you may know Talkswitch requires you to add different accounts for &lt;br /&gt;
 incoming. You can use the sub accounts in this section, however on the customer portal you will need to set the sub accounts for IP &lt;br /&gt;
 authentication and you will have to contact us in order to set all the sub accounts to the same IP address (currently the configuration &lt;br /&gt;
 will require different IP addresses).&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/TalkSwitch</id>
		<title>TalkSwitch</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/TalkSwitch"/>
				<updated>2014-08-29T15:18:20Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt; '''Disclaimer:''' Talkswitch Fortivoice System usage is not 100% guaranteed, the following configuration is the &lt;br /&gt;
 result of several real test with our clients, this is by far the most usable configuration that we have &lt;br /&gt;
 been able to find out, please follow the instructions as explained, and contact us if you have any doubts.&lt;br /&gt;
&lt;br /&gt;
==='''Configuring TalkSwitch for VoIP.ms VoIP Service'''===&lt;br /&gt;
&lt;br /&gt;
This guide will show you how to set up VoIP.ms VoIP service.&lt;br /&gt;
&lt;br /&gt;
When you open an account with VoIP.ms, we will provide you with account credentials information. Use this information to set up the service provider profile and VoIP numbers on your TalkSwitch system.&lt;br /&gt;
&lt;br /&gt;
''Setting up a service provider profile'' &lt;br /&gt;
&lt;br /&gt;
1. Select the VoIP Configuration page. &lt;br /&gt;
&lt;br /&gt;
[[File:Talkswitch1.png]]&lt;br /&gt;
&lt;br /&gt;
2. Select a Profile (SP 1 to SP 4) that you wish to assign for FortiCall.&lt;br /&gt;
&lt;br /&gt;
3. Select the Activate Profile checkbox.&lt;br /&gt;
&lt;br /&gt;
4. Under Service Provider select Manual Setup.&lt;br /&gt;
&lt;br /&gt;
5. Click the Update Configuration button. The essential settings will be completed automatically.&lt;br /&gt;
&lt;br /&gt;
Disable public IP address substitution leave this unchecked.&lt;br /&gt;
Check Register with authentication username.&lt;br /&gt;
Check Enable NAT Keep alives (You can keep the default NAT settings)&lt;br /&gt;
Codec Options: Here make sure to select only the codec supported for VoIP.ms (g.729 and g.711u). And if you have enough bandwidth you should select the g.711u as Preferred codec.&lt;br /&gt;
&lt;br /&gt;
[[File:Image002.png]]&lt;br /&gt;
&lt;br /&gt;
===Provisioning Details===&lt;br /&gt;
*'''Proxy Server name''': server.voip.ms (one of the voip.ms servers, e.g. atlanta.voip.ms)&lt;br /&gt;
*'''Registrar server name''': same as above&lt;br /&gt;
*'''Outbound proxy''': same as above&lt;br /&gt;
*'''Realm/domain''': same as above&lt;br /&gt;
&lt;br /&gt;
==Configuring Outgoing calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
*Select the '''VoIP Numbers''' page&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number outbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
*Select one of the empty voip number slots and then check the '''Activate VoIP number'''&lt;br /&gt;
*'''Select a VoIP profile''': Here select the profile to create for VoIP.ms&lt;br /&gt;
&lt;br /&gt;
'''''The record for your main account sip userID as shown in the screen, this is required as this section sends the correct information required from our servers on the INVITES:FROM which will be the userid, if you don't add this record the Talkswitch will be sending the number (DID set) instead and this will result in continuous 'deregistrations', registration dropping.'''''&lt;br /&gt;
&lt;br /&gt;
===Username and Password===&lt;br /&gt;
&lt;br /&gt;
*'''User/Account''': 100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)&lt;br /&gt;
&lt;br /&gt;
*'''Password''': (the password for your account. If you're using the main account this is the same you use to login to your Customer Portal, unless you change the password, if using a sub account you'll notice each sub account has their own password)&lt;br /&gt;
&lt;br /&gt;
 '''Note''' You can only use the main account for outgoing calls, this is due to the Talkswitch system not supporting&lt;br /&gt;
 the Underscore on the sub accounts.&lt;br /&gt;
&lt;br /&gt;
==Configuring Incoming calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' You don't need to purchase a DID number for outgoing calls. The VoIP number is used as your CallerID number.&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number inbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
Here you can set the callerID number that you want (or your DID number if you want to receive incoming calls), you need to enter the number filling in the 3 fields. &lt;br /&gt;
*'''Country Code''': 1 &lt;br /&gt;
*'''City or area code''': 555&lt;br /&gt;
*'''Number''': 4443322&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[Category:PBXes]]&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' If you are planning to use more than 1 number with us, as you may know Talkswitch requires you to add different accounts for &lt;br /&gt;
 incoming. You can use the sub accounts in this section, however on the customer portal you will need to set the sub accounts for IP &lt;br /&gt;
 authentication and you will have to contact us in order to set all the sub accounts to the same IP address (currently the configuration &lt;br /&gt;
 will require different IP addresses).&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/TalkSwitch</id>
		<title>TalkSwitch</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/TalkSwitch"/>
				<updated>2014-08-29T15:18:06Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt; '''Disclaimer:''' Talkswitch Fortivoice System usage is not 100% guaranteed, the following configuration is the &lt;br /&gt;
 result of several real test with our clients, this is by far the most usable configuration that we have &lt;br /&gt;
 been able to find out, please follow the instructions as explained, and contact us if you have any doubts.&lt;br /&gt;
 Thanks&lt;br /&gt;
&lt;br /&gt;
==='''Configuring TalkSwitch for VoIP.ms VoIP Service'''===&lt;br /&gt;
&lt;br /&gt;
This guide will show you how to set up VoIP.ms VoIP service.&lt;br /&gt;
&lt;br /&gt;
When you open an account with VoIP.ms, we will provide you with account credentials information. Use this information to set up the service provider profile and VoIP numbers on your TalkSwitch system.&lt;br /&gt;
&lt;br /&gt;
''Setting up a service provider profile'' &lt;br /&gt;
&lt;br /&gt;
1. Select the VoIP Configuration page. &lt;br /&gt;
&lt;br /&gt;
[[File:Talkswitch1.png]]&lt;br /&gt;
&lt;br /&gt;
2. Select a Profile (SP 1 to SP 4) that you wish to assign for FortiCall.&lt;br /&gt;
&lt;br /&gt;
3. Select the Activate Profile checkbox.&lt;br /&gt;
&lt;br /&gt;
4. Under Service Provider select Manual Setup.&lt;br /&gt;
&lt;br /&gt;
5. Click the Update Configuration button. The essential settings will be completed automatically.&lt;br /&gt;
&lt;br /&gt;
Disable public IP address substitution leave this unchecked.&lt;br /&gt;
Check Register with authentication username.&lt;br /&gt;
Check Enable NAT Keep alives (You can keep the default NAT settings)&lt;br /&gt;
Codec Options: Here make sure to select only the codec supported for VoIP.ms (g.729 and g.711u). And if you have enough bandwidth you should select the g.711u as Preferred codec.&lt;br /&gt;
&lt;br /&gt;
[[File:Image002.png]]&lt;br /&gt;
&lt;br /&gt;
===Provisioning Details===&lt;br /&gt;
*'''Proxy Server name''': server.voip.ms (one of the voip.ms servers, e.g. atlanta.voip.ms)&lt;br /&gt;
*'''Registrar server name''': same as above&lt;br /&gt;
*'''Outbound proxy''': same as above&lt;br /&gt;
*'''Realm/domain''': same as above&lt;br /&gt;
&lt;br /&gt;
==Configuring Outgoing calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
*Select the '''VoIP Numbers''' page&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number outbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
*Select one of the empty voip number slots and then check the '''Activate VoIP number'''&lt;br /&gt;
*'''Select a VoIP profile''': Here select the profile to create for VoIP.ms&lt;br /&gt;
&lt;br /&gt;
'''''The record for your main account sip userID as shown in the screen, this is required as this section sends the correct information required from our servers on the INVITES:FROM which will be the userid, if you don't add this record the Talkswitch will be sending the number (DID set) instead and this will result in continuous 'deregistrations', registration dropping.'''''&lt;br /&gt;
&lt;br /&gt;
===Username and Password===&lt;br /&gt;
&lt;br /&gt;
*'''User/Account''': 100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)&lt;br /&gt;
&lt;br /&gt;
*'''Password''': (the password for your account. If you're using the main account this is the same you use to login to your Customer Portal, unless you change the password, if using a sub account you'll notice each sub account has their own password)&lt;br /&gt;
&lt;br /&gt;
 '''Note''' You can only use the main account for outgoing calls, this is due to the Talkswitch system not supporting&lt;br /&gt;
 the Underscore on the sub accounts.&lt;br /&gt;
&lt;br /&gt;
==Configuring Incoming calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' You don't need to purchase a DID number for outgoing calls. The VoIP number is used as your CallerID number.&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number inbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
Here you can set the callerID number that you want (or your DID number if you want to receive incoming calls), you need to enter the number filling in the 3 fields. &lt;br /&gt;
*'''Country Code''': 1 &lt;br /&gt;
*'''City or area code''': 555&lt;br /&gt;
*'''Number''': 4443322&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[Category:PBXes]]&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' If you are planning to use more than 1 number with us, as you may know Talkswitch requires you to add different accounts for &lt;br /&gt;
 incoming. You can use the sub accounts in this section, however on the customer portal you will need to set the sub accounts for IP &lt;br /&gt;
 authentication and you will have to contact us in order to set all the sub accounts to the same IP address (currently the configuration &lt;br /&gt;
 will require different IP addresses).&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/TalkSwitch</id>
		<title>TalkSwitch</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/TalkSwitch"/>
				<updated>2014-08-29T15:17:55Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt; '''Disclaimer:''' Talkswitch Fortivoice System usage is not 100% guaranteed, the following configuration is the &lt;br /&gt;
 result of several real test with our clients, this is by far the most usable configuration that we have &lt;br /&gt;
 been able to find out, please follow the instructions as explained, and contact us if you have any doubts.&lt;br /&gt;
&lt;br /&gt;
 Thanks&lt;br /&gt;
&lt;br /&gt;
==='''Configuring TalkSwitch for VoIP.ms VoIP Service'''===&lt;br /&gt;
&lt;br /&gt;
This guide will show you how to set up VoIP.ms VoIP service.&lt;br /&gt;
&lt;br /&gt;
When you open an account with VoIP.ms, we will provide you with account credentials information. Use this information to set up the service provider profile and VoIP numbers on your TalkSwitch system.&lt;br /&gt;
&lt;br /&gt;
''Setting up a service provider profile'' &lt;br /&gt;
&lt;br /&gt;
1. Select the VoIP Configuration page. &lt;br /&gt;
&lt;br /&gt;
[[File:Talkswitch1.png]]&lt;br /&gt;
&lt;br /&gt;
2. Select a Profile (SP 1 to SP 4) that you wish to assign for FortiCall.&lt;br /&gt;
&lt;br /&gt;
3. Select the Activate Profile checkbox.&lt;br /&gt;
&lt;br /&gt;
4. Under Service Provider select Manual Setup.&lt;br /&gt;
&lt;br /&gt;
5. Click the Update Configuration button. The essential settings will be completed automatically.&lt;br /&gt;
&lt;br /&gt;
Disable public IP address substitution leave this unchecked.&lt;br /&gt;
Check Register with authentication username.&lt;br /&gt;
Check Enable NAT Keep alives (You can keep the default NAT settings)&lt;br /&gt;
Codec Options: Here make sure to select only the codec supported for VoIP.ms (g.729 and g.711u). And if you have enough bandwidth you should select the g.711u as Preferred codec.&lt;br /&gt;
&lt;br /&gt;
[[File:Image002.png]]&lt;br /&gt;
&lt;br /&gt;
===Provisioning Details===&lt;br /&gt;
*'''Proxy Server name''': server.voip.ms (one of the voip.ms servers, e.g. atlanta.voip.ms)&lt;br /&gt;
*'''Registrar server name''': same as above&lt;br /&gt;
*'''Outbound proxy''': same as above&lt;br /&gt;
*'''Realm/domain''': same as above&lt;br /&gt;
&lt;br /&gt;
==Configuring Outgoing calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
*Select the '''VoIP Numbers''' page&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number outbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
*Select one of the empty voip number slots and then check the '''Activate VoIP number'''&lt;br /&gt;
*'''Select a VoIP profile''': Here select the profile to create for VoIP.ms&lt;br /&gt;
&lt;br /&gt;
'''''The record for your main account sip userID as shown in the screen, this is required as this section sends the correct information required from our servers on the INVITES:FROM which will be the userid, if you don't add this record the Talkswitch will be sending the number (DID set) instead and this will result in continuous 'deregistrations', registration dropping.'''''&lt;br /&gt;
&lt;br /&gt;
===Username and Password===&lt;br /&gt;
&lt;br /&gt;
*'''User/Account''': 100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)&lt;br /&gt;
&lt;br /&gt;
*'''Password''': (the password for your account. If you're using the main account this is the same you use to login to your Customer Portal, unless you change the password, if using a sub account you'll notice each sub account has their own password)&lt;br /&gt;
&lt;br /&gt;
 '''Note''' You can only use the main account for outgoing calls, this is due to the Talkswitch system not supporting&lt;br /&gt;
 the Underscore on the sub accounts.&lt;br /&gt;
&lt;br /&gt;
==Configuring Incoming calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' You don't need to purchase a DID number for outgoing calls. The VoIP number is used as your CallerID number.&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number inbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
Here you can set the callerID number that you want (or your DID number if you want to receive incoming calls), you need to enter the number filling in the 3 fields. &lt;br /&gt;
*'''Country Code''': 1 &lt;br /&gt;
*'''City or area code''': 555&lt;br /&gt;
*'''Number''': 4443322&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[Category:PBXes]]&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' If you are planning to use more than 1 number with us, as you may know Talkswitch requires you to add different accounts for &lt;br /&gt;
 incoming. You can use the sub accounts in this section, however on the customer portal you will need to set the sub accounts for IP &lt;br /&gt;
 authentication and you will have to contact us in order to set all the sub accounts to the same IP address (currently the configuration &lt;br /&gt;
 will require different IP addresses).&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/TalkSwitch</id>
		<title>TalkSwitch</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/TalkSwitch"/>
				<updated>2014-08-29T15:17:42Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt; '''Disclaimer:''' Talkswitch Fortivoice System usage is not 100% guaranteed, the following configuration is the &lt;br /&gt;
 result of several real test with our clients, this is by far the most usable configuration that we have &lt;br /&gt;
 been able to find out, please follow the instructions as explained, and contact us if you have any doubts. Thanks&lt;br /&gt;
&lt;br /&gt;
==='''Configuring TalkSwitch for VoIP.ms VoIP Service'''===&lt;br /&gt;
&lt;br /&gt;
This guide will show you how to set up VoIP.ms VoIP service.&lt;br /&gt;
&lt;br /&gt;
When you open an account with VoIP.ms, we will provide you with account credentials information. Use this information to set up the service provider profile and VoIP numbers on your TalkSwitch system.&lt;br /&gt;
&lt;br /&gt;
''Setting up a service provider profile'' &lt;br /&gt;
&lt;br /&gt;
1. Select the VoIP Configuration page. &lt;br /&gt;
&lt;br /&gt;
[[File:Talkswitch1.png]]&lt;br /&gt;
&lt;br /&gt;
2. Select a Profile (SP 1 to SP 4) that you wish to assign for FortiCall.&lt;br /&gt;
&lt;br /&gt;
3. Select the Activate Profile checkbox.&lt;br /&gt;
&lt;br /&gt;
4. Under Service Provider select Manual Setup.&lt;br /&gt;
&lt;br /&gt;
5. Click the Update Configuration button. The essential settings will be completed automatically.&lt;br /&gt;
&lt;br /&gt;
Disable public IP address substitution leave this unchecked.&lt;br /&gt;
Check Register with authentication username.&lt;br /&gt;
Check Enable NAT Keep alives (You can keep the default NAT settings)&lt;br /&gt;
Codec Options: Here make sure to select only the codec supported for VoIP.ms (g.729 and g.711u). And if you have enough bandwidth you should select the g.711u as Preferred codec.&lt;br /&gt;
&lt;br /&gt;
[[File:Image002.png]]&lt;br /&gt;
&lt;br /&gt;
===Provisioning Details===&lt;br /&gt;
*'''Proxy Server name''': server.voip.ms (one of the voip.ms servers, e.g. atlanta.voip.ms)&lt;br /&gt;
*'''Registrar server name''': same as above&lt;br /&gt;
*'''Outbound proxy''': same as above&lt;br /&gt;
*'''Realm/domain''': same as above&lt;br /&gt;
&lt;br /&gt;
==Configuring Outgoing calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
*Select the '''VoIP Numbers''' page&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number outbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
*Select one of the empty voip number slots and then check the '''Activate VoIP number'''&lt;br /&gt;
*'''Select a VoIP profile''': Here select the profile to create for VoIP.ms&lt;br /&gt;
&lt;br /&gt;
'''''The record for your main account sip userID as shown in the screen, this is required as this section sends the correct information required from our servers on the INVITES:FROM which will be the userid, if you don't add this record the Talkswitch will be sending the number (DID set) instead and this will result in continuous 'deregistrations', registration dropping.'''''&lt;br /&gt;
&lt;br /&gt;
===Username and Password===&lt;br /&gt;
&lt;br /&gt;
*'''User/Account''': 100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)&lt;br /&gt;
&lt;br /&gt;
*'''Password''': (the password for your account. If you're using the main account this is the same you use to login to your Customer Portal, unless you change the password, if using a sub account you'll notice each sub account has their own password)&lt;br /&gt;
&lt;br /&gt;
 '''Note''' You can only use the main account for outgoing calls, this is due to the Talkswitch system not supporting&lt;br /&gt;
 the Underscore on the sub accounts.&lt;br /&gt;
&lt;br /&gt;
==Configuring Incoming calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' You don't need to purchase a DID number for outgoing calls. The VoIP number is used as your CallerID number.&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number inbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
Here you can set the callerID number that you want (or your DID number if you want to receive incoming calls), you need to enter the number filling in the 3 fields. &lt;br /&gt;
*'''Country Code''': 1 &lt;br /&gt;
*'''City or area code''': 555&lt;br /&gt;
*'''Number''': 4443322&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[Category:PBXes]]&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' If you are planning to use more than 1 number with us, as you may know Talkswitch requires you to add different accounts for &lt;br /&gt;
 incoming. You can use the sub accounts in this section, however on the customer portal you will need to set the sub accounts for IP &lt;br /&gt;
 authentication and you will have to contact us in order to set all the sub accounts to the same IP address (currently the configuration &lt;br /&gt;
 will require different IP addresses).&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/TalkSwitch</id>
		<title>TalkSwitch</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/TalkSwitch"/>
				<updated>2014-08-29T15:17:18Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt; '''Disclaimer:''' Talkswitch Fortivoice System usage is not 100% guaranteed, the following configuration is the &lt;br /&gt;
 result of several real test with our clients, this is by far the most usable configuration that we have been able&lt;br /&gt;
 to find out, please follow the instructions as explained, and contact us if you have any doubts. Thanks&lt;br /&gt;
&lt;br /&gt;
==='''Configuring TalkSwitch for VoIP.ms VoIP Service'''===&lt;br /&gt;
&lt;br /&gt;
This guide will show you how to set up VoIP.ms VoIP service.&lt;br /&gt;
&lt;br /&gt;
When you open an account with VoIP.ms, we will provide you with account credentials information. Use this information to set up the service provider profile and VoIP numbers on your TalkSwitch system.&lt;br /&gt;
&lt;br /&gt;
''Setting up a service provider profile'' &lt;br /&gt;
&lt;br /&gt;
1. Select the VoIP Configuration page. &lt;br /&gt;
&lt;br /&gt;
[[File:Talkswitch1.png]]&lt;br /&gt;
&lt;br /&gt;
2. Select a Profile (SP 1 to SP 4) that you wish to assign for FortiCall.&lt;br /&gt;
&lt;br /&gt;
3. Select the Activate Profile checkbox.&lt;br /&gt;
&lt;br /&gt;
4. Under Service Provider select Manual Setup.&lt;br /&gt;
&lt;br /&gt;
5. Click the Update Configuration button. The essential settings will be completed automatically.&lt;br /&gt;
&lt;br /&gt;
Disable public IP address substitution leave this unchecked.&lt;br /&gt;
Check Register with authentication username.&lt;br /&gt;
Check Enable NAT Keep alives (You can keep the default NAT settings)&lt;br /&gt;
Codec Options: Here make sure to select only the codec supported for VoIP.ms (g.729 and g.711u). And if you have enough bandwidth you should select the g.711u as Preferred codec.&lt;br /&gt;
&lt;br /&gt;
[[File:Image002.png]]&lt;br /&gt;
&lt;br /&gt;
===Provisioning Details===&lt;br /&gt;
*'''Proxy Server name''': server.voip.ms (one of the voip.ms servers, e.g. atlanta.voip.ms)&lt;br /&gt;
*'''Registrar server name''': same as above&lt;br /&gt;
*'''Outbound proxy''': same as above&lt;br /&gt;
*'''Realm/domain''': same as above&lt;br /&gt;
&lt;br /&gt;
==Configuring Outgoing calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
*Select the '''VoIP Numbers''' page&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number outbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
*Select one of the empty voip number slots and then check the '''Activate VoIP number'''&lt;br /&gt;
*'''Select a VoIP profile''': Here select the profile to create for VoIP.ms&lt;br /&gt;
&lt;br /&gt;
'''''The record for your main account sip userID as shown in the screen, this is required as this section sends the correct information required from our servers on the INVITES:FROM which will be the userid, if you don't add this record the Talkswitch will be sending the number (DID set) instead and this will result in continuous 'deregistrations', registration dropping.'''''&lt;br /&gt;
&lt;br /&gt;
===Username and Password===&lt;br /&gt;
&lt;br /&gt;
*'''User/Account''': 100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)&lt;br /&gt;
&lt;br /&gt;
*'''Password''': (the password for your account. If you're using the main account this is the same you use to login to your Customer Portal, unless you change the password, if using a sub account you'll notice each sub account has their own password)&lt;br /&gt;
&lt;br /&gt;
 '''Note''' You can only use the main account for outgoing calls, this is due to the Talkswitch system not supporting&lt;br /&gt;
 the Underscore on the sub accounts.&lt;br /&gt;
&lt;br /&gt;
==Configuring Incoming calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' You don't need to purchase a DID number for outgoing calls. The VoIP number is used as your CallerID number.&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number inbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
Here you can set the callerID number that you want (or your DID number if you want to receive incoming calls), you need to enter the number filling in the 3 fields. &lt;br /&gt;
*'''Country Code''': 1 &lt;br /&gt;
*'''City or area code''': 555&lt;br /&gt;
*'''Number''': 4443322&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[Category:PBXes]]&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' If you are planning to use more than 1 number with us, as you may know Talkswitch requires you to add different accounts for &lt;br /&gt;
 incoming. You can use the sub accounts in this section, however on the customer portal you will need to set the sub accounts for IP &lt;br /&gt;
 authentication and you will have to contact us in order to set all the sub accounts to the same IP address (currently the configuration &lt;br /&gt;
 will require different IP addresses).&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/TalkSwitch</id>
		<title>TalkSwitch</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/TalkSwitch"/>
				<updated>2014-08-29T15:12:45Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt; ''''''Disclaimer'''''' &lt;br /&gt;
&lt;br /&gt;
==='''Configuring TalkSwitch for VoIP.ms VoIP Service'''===&lt;br /&gt;
&lt;br /&gt;
This guide will show you how to set up VoIP.ms VoIP service.&lt;br /&gt;
&lt;br /&gt;
When you open an account with VoIP.ms, we will provide you with account credentials information. Use this information to set up the service provider profile and VoIP numbers on your TalkSwitch system.&lt;br /&gt;
&lt;br /&gt;
''Setting up a service provider profile'' &lt;br /&gt;
&lt;br /&gt;
1. Select the VoIP Configuration page. &lt;br /&gt;
&lt;br /&gt;
[[File:Talkswitch1.png]]&lt;br /&gt;
&lt;br /&gt;
2. Select a Profile (SP 1 to SP 4) that you wish to assign for FortiCall.&lt;br /&gt;
&lt;br /&gt;
3. Select the Activate Profile checkbox.&lt;br /&gt;
&lt;br /&gt;
4. Under Service Provider select Manual Setup.&lt;br /&gt;
&lt;br /&gt;
5. Click the Update Configuration button. The essential settings will be completed automatically.&lt;br /&gt;
&lt;br /&gt;
Disable public IP address substitution leave this unchecked.&lt;br /&gt;
Check Register with authentication username.&lt;br /&gt;
Check Enable NAT Keep alives (You can keep the default NAT settings)&lt;br /&gt;
Codec Options: Here make sure to select only the codec supported for VoIP.ms (g.729 and g.711u). And if you have enough bandwidth you should select the g.711u as Preferred codec.&lt;br /&gt;
&lt;br /&gt;
[[File:Image002.png]]&lt;br /&gt;
&lt;br /&gt;
===Provisioning Details===&lt;br /&gt;
*'''Proxy Server name''': server.voip.ms (one of the voip.ms servers, e.g. atlanta.voip.ms)&lt;br /&gt;
*'''Registrar server name''': same as above&lt;br /&gt;
*'''Outbound proxy''': same as above&lt;br /&gt;
*'''Realm/domain''': same as above&lt;br /&gt;
&lt;br /&gt;
==Configuring Outgoing calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
*Select the '''VoIP Numbers''' page&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number outbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
*Select one of the empty voip number slots and then check the '''Activate VoIP number'''&lt;br /&gt;
*'''Select a VoIP profile''': Here select the profile to create for VoIP.ms&lt;br /&gt;
&lt;br /&gt;
'''''The record for your main account sip userID as shown in the screen, this is required as this section sends the correct information required from our servers on the INVITES:FROM which will be the userid, if you don't add this record the Talkswitch will be sending the number (DID set) instead and this will result in continuous 'deregistrations', registration dropping.'''''&lt;br /&gt;
&lt;br /&gt;
===Username and Password===&lt;br /&gt;
&lt;br /&gt;
*'''User/Account''': 100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)&lt;br /&gt;
&lt;br /&gt;
*'''Password''': (the password for your account. If you're using the main account this is the same you use to login to your Customer Portal, unless you change the password, if using a sub account you'll notice each sub account has their own password)&lt;br /&gt;
&lt;br /&gt;
 '''Note''' You can only use the main account for outgoing calls, this is due to the Talkswitch system not supporting&lt;br /&gt;
 the Underscore on the sub accounts.&lt;br /&gt;
&lt;br /&gt;
==Configuring Incoming calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' You don't need to purchase a DID number for outgoing calls. The VoIP number is used as your CallerID number.&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number inbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
Here you can set the callerID number that you want (or your DID number if you want to receive incoming calls), you need to enter the number filling in the 3 fields. &lt;br /&gt;
*'''Country Code''': 1 &lt;br /&gt;
*'''City or area code''': 555&lt;br /&gt;
*'''Number''': 4443322&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[Category:PBXes]]&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' If you are planning to use more than 1 number with us, as you may know Talkswitch requires you to add different accounts for &lt;br /&gt;
 incoming. You can use the sub accounts in this section, however on the customer portal you will need to set the sub accounts for IP &lt;br /&gt;
 authentication and you will have to contact us in order to set all the sub accounts to the same IP address (currently the configuration &lt;br /&gt;
 will require different IP addresses).&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/TalkSwitch</id>
		<title>TalkSwitch</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/TalkSwitch"/>
				<updated>2014-08-29T15:12:06Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt; '''Disclaimer'''&lt;br /&gt;
&lt;br /&gt;
==='''Configuring TalkSwitch for VoIP.ms VoIP Service'''===&lt;br /&gt;
&lt;br /&gt;
This guide will show you how to set up VoIP.ms VoIP service.&lt;br /&gt;
&lt;br /&gt;
When you open an account with VoIP.ms, we will provide you with account credentials information. Use this information to set up the service provider profile and VoIP numbers on your TalkSwitch system.&lt;br /&gt;
&lt;br /&gt;
''Setting up a service provider profile'' &lt;br /&gt;
&lt;br /&gt;
1. Select the VoIP Configuration page. &lt;br /&gt;
&lt;br /&gt;
[[File:Talkswitch1.png]]&lt;br /&gt;
&lt;br /&gt;
2. Select a Profile (SP 1 to SP 4) that you wish to assign for FortiCall.&lt;br /&gt;
&lt;br /&gt;
3. Select the Activate Profile checkbox.&lt;br /&gt;
&lt;br /&gt;
4. Under Service Provider select Manual Setup.&lt;br /&gt;
&lt;br /&gt;
5. Click the Update Configuration button. The essential settings will be completed automatically.&lt;br /&gt;
&lt;br /&gt;
Disable public IP address substitution leave this unchecked.&lt;br /&gt;
Check Register with authentication username.&lt;br /&gt;
Check Enable NAT Keep alives (You can keep the default NAT settings)&lt;br /&gt;
Codec Options: Here make sure to select only the codec supported for VoIP.ms (g.729 and g.711u). And if you have enough bandwidth you should select the g.711u as Preferred codec.&lt;br /&gt;
&lt;br /&gt;
[[File:Image002.png]]&lt;br /&gt;
&lt;br /&gt;
===Provisioning Details===&lt;br /&gt;
*'''Proxy Server name''': server.voip.ms (one of the voip.ms servers, e.g. atlanta.voip.ms)&lt;br /&gt;
*'''Registrar server name''': same as above&lt;br /&gt;
*'''Outbound proxy''': same as above&lt;br /&gt;
*'''Realm/domain''': same as above&lt;br /&gt;
&lt;br /&gt;
==Configuring Outgoing calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
*Select the '''VoIP Numbers''' page&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number outbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
*Select one of the empty voip number slots and then check the '''Activate VoIP number'''&lt;br /&gt;
*'''Select a VoIP profile''': Here select the profile to create for VoIP.ms&lt;br /&gt;
&lt;br /&gt;
'''''The record for your main account sip userID as shown in the screen, this is required as this section sends the correct information required from our servers on the INVITES:FROM which will be the userid, if you don't add this record the Talkswitch will be sending the number (DID set) instead and this will result in continuous 'deregistrations', registration dropping.'''''&lt;br /&gt;
&lt;br /&gt;
===Username and Password===&lt;br /&gt;
&lt;br /&gt;
*'''User/Account''': 100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)&lt;br /&gt;
&lt;br /&gt;
*'''Password''': (the password for your account. If you're using the main account this is the same you use to login to your Customer Portal, unless you change the password, if using a sub account you'll notice each sub account has their own password)&lt;br /&gt;
&lt;br /&gt;
 '''Note''' You can only use the main account for outgoing calls, this is due to the Talkswitch system not supporting&lt;br /&gt;
 the Underscore on the sub accounts.&lt;br /&gt;
&lt;br /&gt;
==Configuring Incoming calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' You don't need to purchase a DID number for outgoing calls. The VoIP number is used as your CallerID number.&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number inbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
Here you can set the callerID number that you want (or your DID number if you want to receive incoming calls), you need to enter the number filling in the 3 fields. &lt;br /&gt;
*'''Country Code''': 1 &lt;br /&gt;
*'''City or area code''': 555&lt;br /&gt;
*'''Number''': 4443322&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[Category:PBXes]]&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' If you are planning to use more than 1 number with us, as you may know Talkswitch requires you to add different accounts for &lt;br /&gt;
 incoming. You can use the sub accounts in this section, however on the customer portal you will need to set the sub accounts for IP &lt;br /&gt;
 authentication and you will have to contact us in order to set all the sub accounts to the same IP address (currently the configuration &lt;br /&gt;
 will require different IP addresses).&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/TalkSwitch</id>
		<title>TalkSwitch</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/TalkSwitch"/>
				<updated>2014-08-28T16:46:07Z</updated>
		
		<summary type="html">&lt;p&gt;William: /* Username and Password */&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;==='''Configuring TalkSwitch for VoIP.ms VoIP Service'''===&lt;br /&gt;
&lt;br /&gt;
This guide will show you how to set up VoIP.ms VoIP service.&lt;br /&gt;
&lt;br /&gt;
When you start an account with VoIP.ms, we will provide you with account credentials information. Use this information to set up the service provider profile and VoIP numbers on your TalkSwitch system.&lt;br /&gt;
&lt;br /&gt;
''Setting up a service provider profile'' &lt;br /&gt;
&lt;br /&gt;
1. Select the VoIP Configuration page. &lt;br /&gt;
&lt;br /&gt;
[[File:Talkswitch1.png]]&lt;br /&gt;
&lt;br /&gt;
2. Select a Profile (SP 1 to SP 4) that you wish to assign for FortiCall.&lt;br /&gt;
&lt;br /&gt;
3. Select the Activate Profile checkbox.&lt;br /&gt;
&lt;br /&gt;
4. Under Service Provider select Manual Setup.&lt;br /&gt;
&lt;br /&gt;
5. Click the Update Configuration button. The essential settings will be completed automatically.&lt;br /&gt;
&lt;br /&gt;
Disable public IP address substitution leave this unchecked.&lt;br /&gt;
Check Register with authentication username.&lt;br /&gt;
Check Enable NAT Keep alives (You can keep the default NAT settings)&lt;br /&gt;
Codec Options: Here make sure to select only the codec supported for VoIP.ms (g.729 and g.711u). And if you have enough bandwidht you should select the g.711u as Preferred codec.&lt;br /&gt;
&lt;br /&gt;
[[File:Image002.png]]&lt;br /&gt;
&lt;br /&gt;
===Provisioning Details===&lt;br /&gt;
*'''Proxy Server name''': server.voip.ms (one of the voip.ms servers, e.g. atlanta.voip.ms)&lt;br /&gt;
*'''Registrar server name''': same as above&lt;br /&gt;
*'''Outbound proxy''': same as above&lt;br /&gt;
*'''Realm/domain''': same as above&lt;br /&gt;
&lt;br /&gt;
==Configuring Outgoing calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
*Select the '''VoIP Numbers''' page&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number outbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
*Select one of the empty voip number slots and then check the '''Activate VoIP number'''&lt;br /&gt;
*'''Select a VoIP profile''': Here select the profile create for VoIP.ms&lt;br /&gt;
&lt;br /&gt;
'''''The record for your main account sip userID as showed in the screen, this is required as this section sends the correct information required from our servers on the INVITES:FROM which will be the userid, if you don't add this record the Talkswitch will be sending the number (DID set) instead and this will result on continuous deregistration.'''''&lt;br /&gt;
&lt;br /&gt;
===Username and Password===&lt;br /&gt;
&lt;br /&gt;
*'''User/Account''': 100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)&lt;br /&gt;
&lt;br /&gt;
*'''Password''': (the password for your account. If you're using the main account this is the same you use to login in your Customer Portal, unless you change the password, if using a sub account you'll notice each sub account has their own password)&lt;br /&gt;
&lt;br /&gt;
 '''Note''' You can only use the main account for outgoing calls, this is due to Talkswitch system does not support &lt;br /&gt;
 the Underscore on the sub accounts.&lt;br /&gt;
&lt;br /&gt;
==Configuring Incoming calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' You don't need to purchase a DID number for outgoing calls. The VoIP number is used as your CallerID number.&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number inbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
Here you can set the callerID number that you want (or your DID number if you want to receive incoming calls), you need to enter the number filling the 3 fields. &lt;br /&gt;
*'''Country Code''': 1 &lt;br /&gt;
*'''City or area code''': 555&lt;br /&gt;
*'''Number''': 4443322&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[Category:PBXes]]&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' If you are planning to use more than 1 number with us, as you may know the Talkswitch requires you to add different accounts for &lt;br /&gt;
 incoming,you can use the sub accounts on this section however on the customer portal you will need to set the sub accounts for IP &lt;br /&gt;
 authentication, you will have to contact us in order to set all the sub accounts to the same IP address (currently the configuration &lt;br /&gt;
 will require different IP address).&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/TalkSwitch</id>
		<title>TalkSwitch</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/TalkSwitch"/>
				<updated>2014-08-28T16:45:40Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;==='''Configuring TalkSwitch for VoIP.ms VoIP Service'''===&lt;br /&gt;
&lt;br /&gt;
This guide will show you how to set up VoIP.ms VoIP service.&lt;br /&gt;
&lt;br /&gt;
When you start an account with VoIP.ms, we will provide you with account credentials information. Use this information to set up the service provider profile and VoIP numbers on your TalkSwitch system.&lt;br /&gt;
&lt;br /&gt;
''Setting up a service provider profile'' &lt;br /&gt;
&lt;br /&gt;
1. Select the VoIP Configuration page. &lt;br /&gt;
&lt;br /&gt;
[[File:Talkswitch1.png]]&lt;br /&gt;
&lt;br /&gt;
2. Select a Profile (SP 1 to SP 4) that you wish to assign for FortiCall.&lt;br /&gt;
&lt;br /&gt;
3. Select the Activate Profile checkbox.&lt;br /&gt;
&lt;br /&gt;
4. Under Service Provider select Manual Setup.&lt;br /&gt;
&lt;br /&gt;
5. Click the Update Configuration button. The essential settings will be completed automatically.&lt;br /&gt;
&lt;br /&gt;
Disable public IP address substitution leave this unchecked.&lt;br /&gt;
Check Register with authentication username.&lt;br /&gt;
Check Enable NAT Keep alives (You can keep the default NAT settings)&lt;br /&gt;
Codec Options: Here make sure to select only the codec supported for VoIP.ms (g.729 and g.711u). And if you have enough bandwidht you should select the g.711u as Preferred codec.&lt;br /&gt;
&lt;br /&gt;
[[File:Image002.png]]&lt;br /&gt;
&lt;br /&gt;
===Provisioning Details===&lt;br /&gt;
*'''Proxy Server name''': server.voip.ms (one of the voip.ms servers, e.g. atlanta.voip.ms)&lt;br /&gt;
*'''Registrar server name''': same as above&lt;br /&gt;
*'''Outbound proxy''': same as above&lt;br /&gt;
*'''Realm/domain''': same as above&lt;br /&gt;
&lt;br /&gt;
==Configuring Outgoing calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
*Select the '''VoIP Numbers''' page&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number outbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
*Select one of the empty voip number slots and then check the '''Activate VoIP number'''&lt;br /&gt;
*'''Select a VoIP profile''': Here select the profile create for VoIP.ms&lt;br /&gt;
&lt;br /&gt;
'''''The record for your main account sip userID as showed in the screen, this is required as this section sends the correct information required from our servers on the INVITES:FROM which will be the userid, if you don't add this record the Talkswitch will be sending the number (DID set) instead and this will result on continuous deregistration.'''''&lt;br /&gt;
&lt;br /&gt;
===Username and Password===&lt;br /&gt;
&lt;br /&gt;
*'''User/Account''': 100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)&lt;br /&gt;
&lt;br /&gt;
*'''Password''': (the password for your account. If you're using the main account this is the same you use to login in your Customer Portal, unless you change the password, if using a sub account you'll notice each sub account has their own password)&lt;br /&gt;
&lt;br /&gt;
 '''Note''' You can only use the main account for outgoing calls, this is due to Talkswitch system does not support the Underscore on the sub accounts. &lt;br /&gt;
&lt;br /&gt;
==Configuring Incoming calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' You don't need to purchase a DID number for outgoing calls. The VoIP number is used as your CallerID number.&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number inbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
Here you can set the callerID number that you want (or your DID number if you want to receive incoming calls), you need to enter the number filling the 3 fields. &lt;br /&gt;
*'''Country Code''': 1 &lt;br /&gt;
*'''City or area code''': 555&lt;br /&gt;
*'''Number''': 4443322&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[Category:PBXes]]&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' If you are planning to use more than 1 number with us, as you may know the Talkswitch requires you to add different accounts for &lt;br /&gt;
 incoming,you can use the sub accounts on this section however on the customer portal you will need to set the sub accounts for IP &lt;br /&gt;
 authentication, you will have to contact us in order to set all the sub accounts to the same IP address (currently the configuration &lt;br /&gt;
 will require different IP address).&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/TalkSwitch</id>
		<title>TalkSwitch</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/TalkSwitch"/>
				<updated>2014-08-28T16:45:22Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;==='''Configuring TalkSwitch for VoIP.ms VoIP Service'''===&lt;br /&gt;
&lt;br /&gt;
This guide will show you how to set up VoIP.ms VoIP service.&lt;br /&gt;
&lt;br /&gt;
When you start an account with VoIP.ms, we will provide you with account credentials information. Use this information to set up the service provider profile and VoIP numbers on your TalkSwitch system.&lt;br /&gt;
&lt;br /&gt;
''Setting up a service provider profile'' &lt;br /&gt;
&lt;br /&gt;
1. Select the VoIP Configuration page. &lt;br /&gt;
&lt;br /&gt;
[[File:Talkswitch1.png]]&lt;br /&gt;
&lt;br /&gt;
2. Select a Profile (SP 1 to SP 4) that you wish to assign for FortiCall.&lt;br /&gt;
&lt;br /&gt;
3. Select the Activate Profile checkbox.&lt;br /&gt;
&lt;br /&gt;
4. Under Service Provider select Manual Setup.&lt;br /&gt;
&lt;br /&gt;
5. Click the Update Configuration button. The essential settings will be completed automatically.&lt;br /&gt;
&lt;br /&gt;
Disable public IP address substitution leave this unchecked.&lt;br /&gt;
Check Register with authentication username.&lt;br /&gt;
Check Enable NAT Keep alives (You can keep the default NAT settings)&lt;br /&gt;
Codec Options: Here make sure to select only the codec supported for VoIP.ms (g.729 and g.711u). And if you have enough bandwidht you should select the g.711u as Preferred codec.&lt;br /&gt;
&lt;br /&gt;
[[File:Image002.png]]&lt;br /&gt;
&lt;br /&gt;
===Provisioning Details===&lt;br /&gt;
*'''Proxy Server name''': server.voip.ms (one of the voip.ms servers, e.g. atlanta.voip.ms)&lt;br /&gt;
*'''Registrar server name''': same as above&lt;br /&gt;
*'''Outbound proxy''': same as above&lt;br /&gt;
*'''Realm/domain''': same as above&lt;br /&gt;
&lt;br /&gt;
==Configuring Outgoing calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
*Select the '''VoIP Numbers''' page&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number outbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
*Select one of the empty voip number slots and then check the '''Activate VoIP number'''&lt;br /&gt;
*'''Select a VoIP profile''': Here select the profile create for VoIP.ms&lt;br /&gt;
&lt;br /&gt;
'''''The record for your main account sip userID as showed in the screen, this is required as this section sends the correct information required from our servers on the INVITES:FROM which will be the userid, if you don't add this record the Talkswitch will be sending the number (DID set) instead and this will result on continuous deregistration.'''''&lt;br /&gt;
&lt;br /&gt;
===Username and Password===&lt;br /&gt;
&lt;br /&gt;
*'''User/Account''': 100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)&lt;br /&gt;
&lt;br /&gt;
*'''Password''': (the password for your account. If you're using the main account this is the same you use to login in your Customer Portal, unless you change the password, if using a sub account you'll notice each sub account has their own password)&lt;br /&gt;
&lt;br /&gt;
 '''Note''' You can only use the main account for outgoing calls, this is due to Talkswitch system does not support the Underscore on the sub accounts. &lt;br /&gt;
&lt;br /&gt;
==Configuring Incoming calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' You don't need to purchase a DID number for outgoing calls. The VoIP number is used as your CallerID number.&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number inbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
Here you can set the callerID number that you want (or your DID number if you want to receive incoming calls), you need to enter the number filling the 3 fields. &lt;br /&gt;
*'''Country Code''': 1 &lt;br /&gt;
*'''City or area code''': 555&lt;br /&gt;
*'''Number''': 4443322&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[Category:PBXes]]&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' If you are planning to use more than 1 number with us, as you may know the Talkswitch requires you to add different accounts for &lt;br /&gt;
 incoming,you can use the sub accounts on this section however on the customer portal you will need to set the sub accounts for IP &lt;br /&gt;
 authentication, you will have to contact us in order to set all the sub accounts to the same IP address (currently the configuration will require different IP address).&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/TalkSwitch</id>
		<title>TalkSwitch</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/TalkSwitch"/>
				<updated>2014-08-28T16:44:56Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;==='''Configuring TalkSwitch for VoIP.ms VoIP Service'''===&lt;br /&gt;
&lt;br /&gt;
This guide will show you how to set up VoIP.ms VoIP service.&lt;br /&gt;
&lt;br /&gt;
When you start an account with VoIP.ms, we will provide you with account credentials information. Use this information to set up the service provider profile and VoIP numbers on your TalkSwitch system.&lt;br /&gt;
&lt;br /&gt;
''Setting up a service provider profile'' &lt;br /&gt;
&lt;br /&gt;
1. Select the VoIP Configuration page. &lt;br /&gt;
&lt;br /&gt;
[[File:Talkswitch1.png]]&lt;br /&gt;
&lt;br /&gt;
2. Select a Profile (SP 1 to SP 4) that you wish to assign for FortiCall.&lt;br /&gt;
&lt;br /&gt;
3. Select the Activate Profile checkbox.&lt;br /&gt;
&lt;br /&gt;
4. Under Service Provider select Manual Setup.&lt;br /&gt;
&lt;br /&gt;
5. Click the Update Configuration button. The essential settings will be completed automatically.&lt;br /&gt;
&lt;br /&gt;
Disable public IP address substitution leave this unchecked.&lt;br /&gt;
Check Register with authentication username.&lt;br /&gt;
Check Enable NAT Keep alives (You can keep the default NAT settings)&lt;br /&gt;
Codec Options: Here make sure to select only the codec supported for VoIP.ms (g.729 and g.711u). And if you have enough bandwidht you should select the g.711u as Preferred codec.&lt;br /&gt;
&lt;br /&gt;
[[File:Image002.png]]&lt;br /&gt;
&lt;br /&gt;
===Provisioning Details===&lt;br /&gt;
*'''Proxy Server name''': server.voip.ms (one of the voip.ms servers, e.g. atlanta.voip.ms)&lt;br /&gt;
*'''Registrar server name''': same as above&lt;br /&gt;
*'''Outbound proxy''': same as above&lt;br /&gt;
*'''Realm/domain''': same as above&lt;br /&gt;
&lt;br /&gt;
==Configuring Outgoing calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
*Select the '''VoIP Numbers''' page&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number outbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
*Select one of the empty voip number slots and then check the '''Activate VoIP number'''&lt;br /&gt;
*'''Select a VoIP profile''': Here select the profile create for VoIP.ms&lt;br /&gt;
&lt;br /&gt;
'''''The record for your main account sip userID as showed in the screen, this is required as this section sends the correct information required from our servers on the INVITES:FROM which will be the userid, if you don't add this record the Talkswitch will be sending the number (DID set) instead and this will result on continuous deregistration.'''''&lt;br /&gt;
&lt;br /&gt;
===Username and Password===&lt;br /&gt;
&lt;br /&gt;
*'''User/Account''': 100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)&lt;br /&gt;
&lt;br /&gt;
*'''Password''': (the password for your account. If you're using the main account this is the same you use to login in your Customer Portal, unless you change the password, if using a sub account you'll notice each sub account has their own password)&lt;br /&gt;
&lt;br /&gt;
 '''Note''' You can only use the main account for outgoing calls, this is due to Talkswitch system does not support the Underscore on the sub accounts. &lt;br /&gt;
&lt;br /&gt;
==Configuring Incoming calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' You don't need to purchase a DID number for outgoing calls. The VoIP number is used as your CallerID number.&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number inbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
Here you can set the callerID number that you want (or your DID number if you want to receive incoming calls), you need to enter the number filling the 3 fields. &lt;br /&gt;
*'''Country Code''': 1 &lt;br /&gt;
*'''City or area code''': 555&lt;br /&gt;
*'''Number''': 4443322&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[Category:PBXes]]&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' If you are planning to use more than 1 number with us, as you may know the Talkswitch requires you to add different accounts for &lt;br /&gt;
 incoming,you can use the sub accounts on this section however on the customer portal you will need to set the sub accounts for IP authentication, you will have to contact us in order &lt;br /&gt;
 to set all the sub accounts to the same IP address (currently the configuration will require different IP address).&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/TalkSwitch</id>
		<title>TalkSwitch</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/TalkSwitch"/>
				<updated>2014-08-28T16:43:37Z</updated>
		
		<summary type="html">&lt;p&gt;William: /* Configuring Outgoing calls with VoIP.ms */&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;==='''Configuring TalkSwitch for VoIP.ms VoIP Service'''===&lt;br /&gt;
&lt;br /&gt;
This guide will show you how to set up VoIP.ms VoIP service.&lt;br /&gt;
&lt;br /&gt;
When you start an account with VoIP.ms, we will provide you with account credentials information. Use this information to set up the service provider profile and VoIP numbers on your TalkSwitch system.&lt;br /&gt;
&lt;br /&gt;
''Setting up a service provider profile'' &lt;br /&gt;
&lt;br /&gt;
1. Select the VoIP Configuration page. &lt;br /&gt;
&lt;br /&gt;
[[File:Talkswitch1.png]]&lt;br /&gt;
&lt;br /&gt;
2. Select a Profile (SP 1 to SP 4) that you wish to assign for FortiCall.&lt;br /&gt;
&lt;br /&gt;
3. Select the Activate Profile checkbox.&lt;br /&gt;
&lt;br /&gt;
4. Under Service Provider select Manual Setup.&lt;br /&gt;
&lt;br /&gt;
5. Click the Update Configuration button. The essential settings will be completed automatically.&lt;br /&gt;
&lt;br /&gt;
Disable public IP address substitution leave this unchecked.&lt;br /&gt;
Check Register with authentication username.&lt;br /&gt;
Check Enable NAT Keep alives (You can keep the default NAT settings)&lt;br /&gt;
Codec Options: Here make sure to select only the codec supported for VoIP.ms (g.729 and g.711u). And if you have enough bandwidht you should select the g.711u as Preferred codec.&lt;br /&gt;
&lt;br /&gt;
[[File:Image002.png]]&lt;br /&gt;
&lt;br /&gt;
===Provisioning Details===&lt;br /&gt;
*'''Proxy Server name''': server.voip.ms (one of the voip.ms servers, e.g. atlanta.voip.ms)&lt;br /&gt;
*'''Registrar server name''': same as above&lt;br /&gt;
*'''Outbound proxy''': same as above&lt;br /&gt;
*'''Realm/domain''': same as above&lt;br /&gt;
&lt;br /&gt;
==Configuring Outgoing calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
*Select the '''VoIP Numbers''' page&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number outbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
*Select one of the empty voip number slots and then check the '''Activate VoIP number'''&lt;br /&gt;
*'''Select a VoIP profile''': Here select the profile create for VoIP.ms&lt;br /&gt;
&lt;br /&gt;
'''''The record for your main account sip userID as showed in the screen, this is required as this section sends the correct information required from our servers on the INVITES:FROM which will be the userid, if you don't add this record the Talkswitch will be sending the number (DID set) instead and this will result on continuous deregistration.'''''&lt;br /&gt;
&lt;br /&gt;
===Username and Password===&lt;br /&gt;
&lt;br /&gt;
*'''User/Account''': 100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)&lt;br /&gt;
&lt;br /&gt;
*'''Password''': (the password for your account. If you're using the main account this is the same you use to login in your Customer Portal, unless you change the password, if using a sub account you'll notice each sub account has their own password)&lt;br /&gt;
&lt;br /&gt;
 '''Note''' You can only use the main account for outgoing calls, this is due to Talkswitch system does not support the Underscore on the sub accounts. &lt;br /&gt;
&lt;br /&gt;
==Configuring Incoming calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' You don't need to purchase a DID number for outgoing calls. The VoIP number is used as your CallerID number.&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number inbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
Here you can set the callerID number that you want (or your DID number if you want to receive incoming calls), you need to enter the number filling the 3 fields. &lt;br /&gt;
*'''Country Code''': 1 &lt;br /&gt;
*'''City or area code''': 555&lt;br /&gt;
*'''Number''': 4443322&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[Category:PBXes]]&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' If you are planning to use more than 1 number with us, as you may know the Talkswitch requires you to add different accounts for incoming, &lt;br /&gt;
&lt;br /&gt;
you can use the sub accounts on this section however on the customer portal you will need to set the sub accounts for IP authentication, you will have to contact us in order to set all the sub accounts to the same IP address (currently the configuration will require different IP address).&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/TalkSwitch</id>
		<title>TalkSwitch</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/TalkSwitch"/>
				<updated>2014-08-28T16:42:47Z</updated>
		
		<summary type="html">&lt;p&gt;William: /* Configuring Outgoing calls with VoIP.ms */&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;==='''Configuring TalkSwitch for VoIP.ms VoIP Service'''===&lt;br /&gt;
&lt;br /&gt;
This guide will show you how to set up VoIP.ms VoIP service.&lt;br /&gt;
&lt;br /&gt;
When you start an account with VoIP.ms, we will provide you with account credentials information. Use this information to set up the service provider profile and VoIP numbers on your TalkSwitch system.&lt;br /&gt;
&lt;br /&gt;
''Setting up a service provider profile'' &lt;br /&gt;
&lt;br /&gt;
1. Select the VoIP Configuration page. &lt;br /&gt;
&lt;br /&gt;
[[File:Talkswitch1.png]]&lt;br /&gt;
&lt;br /&gt;
2. Select a Profile (SP 1 to SP 4) that you wish to assign for FortiCall.&lt;br /&gt;
&lt;br /&gt;
3. Select the Activate Profile checkbox.&lt;br /&gt;
&lt;br /&gt;
4. Under Service Provider select Manual Setup.&lt;br /&gt;
&lt;br /&gt;
5. Click the Update Configuration button. The essential settings will be completed automatically.&lt;br /&gt;
&lt;br /&gt;
Disable public IP address substitution leave this unchecked.&lt;br /&gt;
Check Register with authentication username.&lt;br /&gt;
Check Enable NAT Keep alives (You can keep the default NAT settings)&lt;br /&gt;
Codec Options: Here make sure to select only the codec supported for VoIP.ms (g.729 and g.711u). And if you have enough bandwidht you should select the g.711u as Preferred codec.&lt;br /&gt;
&lt;br /&gt;
[[File:Image002.png]]&lt;br /&gt;
&lt;br /&gt;
===Provisioning Details===&lt;br /&gt;
*'''Proxy Server name''': server.voip.ms (one of the voip.ms servers, e.g. atlanta.voip.ms)&lt;br /&gt;
*'''Registrar server name''': same as above&lt;br /&gt;
*'''Outbound proxy''': same as above&lt;br /&gt;
*'''Realm/domain''': same as above&lt;br /&gt;
&lt;br /&gt;
==Configuring Outgoing calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
*Select the '''VoIP Numbers''' page&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number outbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
*Select one of the empty voip number slots and then check the '''Activate VoIP number'''&lt;br /&gt;
*'''Select a VoIP profile''': Here select the profile create for VoIP.ms&lt;br /&gt;
&lt;br /&gt;
'''''The record for your main account sip userID as showed in the screen, this is required as this section sends the correct information required from our servers on the INVITES:FROM which will be the userid, if you don't add this record the Talkswitch will be sending the number (DID set) instead and this will result on continuous deregistration.'''''&lt;br /&gt;
&lt;br /&gt;
===Username and Password===&lt;br /&gt;
&lt;br /&gt;
*'''User/Account''': 100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)&lt;br /&gt;
&lt;br /&gt;
*'''Password''': (the password for your account. If you're using the main account this is the same you use to login in your Customer Portal, unless you change the password, if using a sub account you'll notice each sub account has their own password)&lt;br /&gt;
&lt;br /&gt;
 '''Note''' You can only use the main account for outgoing calls, this is due to Talkswitch system does not support the Underscore on the sub accounts. &lt;br /&gt;
&lt;br /&gt;
===Configuring Incoming calls with VoIP.ms===&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' You don't need to purchase a DID number for outgoing calls. The VoIP number is used as your CallerID number.&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number inbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
Here you can set the callerID number that you want (or your DID number if you want to receive incoming calls), you need to enter the number filling the 3 fields. &lt;br /&gt;
*'''Country Code''': 1 &lt;br /&gt;
*'''City or area code''': 555&lt;br /&gt;
*'''Number''': 4443322&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[Category:PBXes]]&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' If you are planning to use more than 1 number with us, as you may know the Talkswitch requires you to add different accounts for incoming, &lt;br /&gt;
&lt;br /&gt;
you can use the sub accounts on this section however on the customer portal you will need to set the sub accounts for IP authentication, you will have to contact us in order to set all the sub accounts to the same IP address (currently the configuration will require different IP address).&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/TalkSwitch</id>
		<title>TalkSwitch</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/TalkSwitch"/>
				<updated>2014-08-28T16:42:10Z</updated>
		
		<summary type="html">&lt;p&gt;William: &lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;==='''Configuring TalkSwitch for VoIP.ms VoIP Service'''===&lt;br /&gt;
&lt;br /&gt;
This guide will show you how to set up VoIP.ms VoIP service.&lt;br /&gt;
&lt;br /&gt;
When you start an account with VoIP.ms, we will provide you with account credentials information. Use this information to set up the service provider profile and VoIP numbers on your TalkSwitch system.&lt;br /&gt;
&lt;br /&gt;
''Setting up a service provider profile'' &lt;br /&gt;
&lt;br /&gt;
1. Select the VoIP Configuration page. &lt;br /&gt;
&lt;br /&gt;
[[File:Talkswitch1.png]]&lt;br /&gt;
&lt;br /&gt;
2. Select a Profile (SP 1 to SP 4) that you wish to assign for FortiCall.&lt;br /&gt;
&lt;br /&gt;
3. Select the Activate Profile checkbox.&lt;br /&gt;
&lt;br /&gt;
4. Under Service Provider select Manual Setup.&lt;br /&gt;
&lt;br /&gt;
5. Click the Update Configuration button. The essential settings will be completed automatically.&lt;br /&gt;
&lt;br /&gt;
Disable public IP address substitution leave this unchecked.&lt;br /&gt;
Check Register with authentication username.&lt;br /&gt;
Check Enable NAT Keep alives (You can keep the default NAT settings)&lt;br /&gt;
Codec Options: Here make sure to select only the codec supported for VoIP.ms (g.729 and g.711u). And if you have enough bandwidht you should select the g.711u as Preferred codec.&lt;br /&gt;
&lt;br /&gt;
[[File:Image002.png]]&lt;br /&gt;
&lt;br /&gt;
===Provisioning Details===&lt;br /&gt;
*'''Proxy Server name''': server.voip.ms (one of the voip.ms servers, e.g. atlanta.voip.ms)&lt;br /&gt;
*'''Registrar server name''': same as above&lt;br /&gt;
*'''Outbound proxy''': same as above&lt;br /&gt;
*'''Realm/domain''': same as above&lt;br /&gt;
&lt;br /&gt;
==Configuring Outgoing calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
*Select the '''VoIP Numbers''' page&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number outbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
*Select one of the empty voip number slots and then check the '''Activate VoIP number'''&lt;br /&gt;
*'''Select a VoIP profile''': Here select the profile create for VoIP.ms&lt;br /&gt;
&lt;br /&gt;
'''''Before adding the numbers you will require to create one record for your main account sip userID as showed in the screen, this is required as this section sends the correct information required from our servers on the INVITES:FROM which will be the userid, if you don't add this record the Talkswitch will be sending the number (DID set) instead and this will result on continuous deregistration.'''''&lt;br /&gt;
&lt;br /&gt;
===Username and Password===&lt;br /&gt;
&lt;br /&gt;
*'''User/Account''': 100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)&lt;br /&gt;
&lt;br /&gt;
*'''Password''': (the password for your account. If you're using the main account this is the same you use to login in your Customer Portal, unless you change the password, if using a sub account you'll notice each sub account has their own password)&lt;br /&gt;
&lt;br /&gt;
 '''Note''' You can only use the main account for outgoing calls, this is due to Talkswitch system does not support the Underscore on the sub accounts. &lt;br /&gt;
&lt;br /&gt;
===Configuring Incoming calls with VoIP.ms===&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' You don't need to purchase a DID number for outgoing calls. The VoIP number is used as your CallerID number.&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number inbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
Here you can set the callerID number that you want (or your DID number if you want to receive incoming calls), you need to enter the number filling the 3 fields. &lt;br /&gt;
*'''Country Code''': 1 &lt;br /&gt;
*'''City or area code''': 555&lt;br /&gt;
*'''Number''': 4443322&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
[[Category:PBXes]]&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' If you are planning to use more than 1 number with us, as you may know the Talkswitch requires you to add different accounts for incoming, &lt;br /&gt;
&lt;br /&gt;
you can use the sub accounts on this section however on the customer portal you will need to set the sub accounts for IP authentication, you will have to contact us in order to set all the sub accounts to the same IP address (currently the configuration will require different IP address).&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/TalkSwitch</id>
		<title>TalkSwitch</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/TalkSwitch"/>
				<updated>2014-08-28T16:40:30Z</updated>
		
		<summary type="html">&lt;p&gt;William: /* Phone Number */&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;==='''Configuring TalkSwitch for VoIP.ms VoIP Service'''===&lt;br /&gt;
&lt;br /&gt;
This guide will show you how to set up VoIP.ms VoIP service.&lt;br /&gt;
&lt;br /&gt;
When you start an account with VoIP.ms, we will provide you with account credentials information. Use this information to set up the service provider profile and VoIP numbers on your TalkSwitch system.&lt;br /&gt;
&lt;br /&gt;
''Setting up a service provider profile'' &lt;br /&gt;
&lt;br /&gt;
1. Select the VoIP Configuration page. &lt;br /&gt;
&lt;br /&gt;
[[File:Talkswitch1.png]]&lt;br /&gt;
&lt;br /&gt;
2. Select a Profile (SP 1 to SP 4) that you wish to assign for FortiCall.&lt;br /&gt;
&lt;br /&gt;
3. Select the Activate Profile checkbox.&lt;br /&gt;
&lt;br /&gt;
4. Under Service Provider select Manual Setup.&lt;br /&gt;
&lt;br /&gt;
5. Click the Update Configuration button. The essential settings will be completed automatically.&lt;br /&gt;
&lt;br /&gt;
Disable public IP address substitution leave this unchecked.&lt;br /&gt;
Check Register with authentication username.&lt;br /&gt;
Check Enable NAT Keep alives (You can keep the default NAT settings)&lt;br /&gt;
Codec Options: Here make sure to select only the codec supported for VoIP.ms (g.729 and g.711u). And if you have enough bandwidht you should select the g.711u as Preferred codec.&lt;br /&gt;
&lt;br /&gt;
[[File:Image002.png]]&lt;br /&gt;
&lt;br /&gt;
===Provisioning Details===&lt;br /&gt;
*'''Proxy Server name''': server.voip.ms (one of the voip.ms servers, e.g. atlanta.voip.ms)&lt;br /&gt;
*'''Registrar server name''': same as above&lt;br /&gt;
*'''Outbound proxy''': same as above&lt;br /&gt;
*'''Realm/domain''': same as above&lt;br /&gt;
&lt;br /&gt;
==Configuring Outgoing calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
*Select the '''VoIP Numbers''' page&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number outbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
*Select one of the empty voip number slots and then check the '''Activate VoIP number'''&lt;br /&gt;
*'''Select a VoIP profile''': Here select the profile create for VoIP.ms&lt;br /&gt;
&lt;br /&gt;
'''''Before adding the numbers you will require to create one record for your main account sip userID as showed in the screen, this is required as this section sends the correct information required from our servers on the INVITES:FROM which will be the userid, if you don't add this record the Talkswitch will be sending the number (DID set) instead and this will result on continuous deregistration.'''''&lt;br /&gt;
&lt;br /&gt;
===Configuring Incoming calls with VoIP.ms===&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' You don't need to purchase a DID number for outgoing calls. The VoIP number is used as your CallerID number.&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number inbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
Here you can set the callerID number that you want (or your DID number if you want to receive incoming calls), you need to enter the number filling the 3 fields. &lt;br /&gt;
*'''Country Code''': 1 &lt;br /&gt;
*'''City or area code''': 555&lt;br /&gt;
*'''Number''': 4443322&lt;br /&gt;
&lt;br /&gt;
===Username and Password===&lt;br /&gt;
&lt;br /&gt;
*'''User/Account''': 100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)&lt;br /&gt;
&lt;br /&gt;
*'''Password''': (the password for your account. If you're using the main account this is the same you use to login in your Customer Portal, unless you change the password, if using a sub account you'll notice each sub account has their own password)&lt;br /&gt;
[[Category:PBXes]]&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' If you are planning to use more than 1 number with us, as you may know the Talkswitch requires you to add different accounts for incoming, &lt;br /&gt;
&lt;br /&gt;
you can use the sub accounts on this section however on the customer portal you will need to set the sub accounts for IP authentication, you will have to contact us in order to set all the sub accounts to the same IP address (currently the configuration will require different IP address).&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	<entry>
		<id>https://wiki.voip.ms/article/TalkSwitch</id>
		<title>TalkSwitch</title>
		<link rel="alternate" type="text/html" href="https://wiki.voip.ms/article/TalkSwitch"/>
				<updated>2014-08-28T16:39:51Z</updated>
		
		<summary type="html">&lt;p&gt;William: /* Configuring a VoIP number with VoIP.ms */&lt;/p&gt;
&lt;hr /&gt;
&lt;div&gt;==='''Configuring TalkSwitch for VoIP.ms VoIP Service'''===&lt;br /&gt;
&lt;br /&gt;
This guide will show you how to set up VoIP.ms VoIP service.&lt;br /&gt;
&lt;br /&gt;
When you start an account with VoIP.ms, we will provide you with account credentials information. Use this information to set up the service provider profile and VoIP numbers on your TalkSwitch system.&lt;br /&gt;
&lt;br /&gt;
''Setting up a service provider profile'' &lt;br /&gt;
&lt;br /&gt;
1. Select the VoIP Configuration page. &lt;br /&gt;
&lt;br /&gt;
[[File:Talkswitch1.png]]&lt;br /&gt;
&lt;br /&gt;
2. Select a Profile (SP 1 to SP 4) that you wish to assign for FortiCall.&lt;br /&gt;
&lt;br /&gt;
3. Select the Activate Profile checkbox.&lt;br /&gt;
&lt;br /&gt;
4. Under Service Provider select Manual Setup.&lt;br /&gt;
&lt;br /&gt;
5. Click the Update Configuration button. The essential settings will be completed automatically.&lt;br /&gt;
&lt;br /&gt;
Disable public IP address substitution leave this unchecked.&lt;br /&gt;
Check Register with authentication username.&lt;br /&gt;
Check Enable NAT Keep alives (You can keep the default NAT settings)&lt;br /&gt;
Codec Options: Here make sure to select only the codec supported for VoIP.ms (g.729 and g.711u). And if you have enough bandwidht you should select the g.711u as Preferred codec.&lt;br /&gt;
&lt;br /&gt;
[[File:Image002.png]]&lt;br /&gt;
&lt;br /&gt;
===Provisioning Details===&lt;br /&gt;
*'''Proxy Server name''': server.voip.ms (one of the voip.ms servers, e.g. atlanta.voip.ms)&lt;br /&gt;
*'''Registrar server name''': same as above&lt;br /&gt;
*'''Outbound proxy''': same as above&lt;br /&gt;
*'''Realm/domain''': same as above&lt;br /&gt;
&lt;br /&gt;
==Configuring Outgoing calls with VoIP.ms==&lt;br /&gt;
&lt;br /&gt;
&lt;br /&gt;
*Select the '''VoIP Numbers''' page&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number outbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
*Select one of the empty voip number slots and then check the '''Activate VoIP number'''&lt;br /&gt;
*'''Select a VoIP profile''': Here select the profile create for VoIP.ms&lt;br /&gt;
&lt;br /&gt;
'''''Before adding the numbers you will require to create one record for your main account sip userID as showed in the screen, this is required as this section sends the correct information required from our servers on the INVITES:FROM which will be the userid, if you don't add this record the Talkswitch will be sending the number (DID set) instead and this will result on continuous deregistration.'''''&lt;br /&gt;
&lt;br /&gt;
===Phone Number===&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' You don't need to purchase a DID number for outgoing calls. The VoIP number is used as your CallerID number.&lt;br /&gt;
&lt;br /&gt;
[[File:Voip number inbound (3).jpg|800px]]&lt;br /&gt;
&lt;br /&gt;
Here you can set the callerID number that you want (or your DID number if you want to receive incoming calls), you need to enter the number filling the 3 fields. &lt;br /&gt;
*'''Country Code''': 1 &lt;br /&gt;
*'''City or area code''': 555&lt;br /&gt;
*'''Number''': 4443322&lt;br /&gt;
&lt;br /&gt;
===Username and Password===&lt;br /&gt;
&lt;br /&gt;
*'''User/Account''': 100000 (Replace with your 6 digit Main SIP Account User ID or Sub Account username, i.e. 123456 or 123456_sub)&lt;br /&gt;
&lt;br /&gt;
*'''Password''': (the password for your account. If you're using the main account this is the same you use to login in your Customer Portal, unless you change the password, if using a sub account you'll notice each sub account has their own password)&lt;br /&gt;
[[Category:PBXes]]&lt;br /&gt;
&lt;br /&gt;
 '''Note:''' If you are planning to use more than 1 number with us, as you may know the Talkswitch requires you to add different accounts for incoming, &lt;br /&gt;
&lt;br /&gt;
you can use the sub accounts on this section however on the customer portal you will need to set the sub accounts for IP authentication, you will have to contact us in order to set all the sub accounts to the same IP address (currently the configuration will require different IP address).&lt;/div&gt;</summary>
		<author><name>William</name></author>	</entry>

	</feed>