VoIP.ms Wiki - User contributions [en]

Ring Groups

2014-01-16T16:54:05Z

Ed:

The Ring Group feature allows you to have incoming calls to be redirected to different destinations that are included in your Ring Group, where a member of the group is able to answer. When you receive a call to a DID routed to a Ring Group, all members of that group will ring at the same time until one of them answers the call. You can add various types of members to a ring group:
Main Account,
[[Sub Accounts]],
[[SIP URI]]'s,
[[Call Forwarding]].

You can also select which voicemail should be used by the system in case none of the members answer the call. The limit of members in a Ring Group is 12: Up to 8 (SIP, IAX2, or SIP URI) members and up to 4 call forward entries per each Ring Group.

[[File:RingGroup.jpg|800px]]

== Creating a Ring Group Entry ==
From your main portal please refer to DID Numbers -> Ring Group. You will have to click on the link that reads ''Click here to create a new ring group''. Then another screen will prompt and you will have to enter the following information:

'''Description:''' This can be used as a note or description to easily identify your ring groups.

'''Members:''' Here you can select the members of the ring group, remember that the limit is 8 members.

'''Voicemail:''' Optionally, you can select a voicemail for this ring group that will override the default DID Voicemail.

Finally, just hit the '''Create''' button and you will be done creating your entry.

[[File:RingGroup_Create.jpg|800px]]

== Routing your DID to your Ring Group ==

After you have created your Ring Group entry, you can route any of your DIDs to your Ring Group entry from your main portal. Please refer to DID Numbers -> [[Manage DID]] -> Edit DID -> Routing -> Ring Group. Also under the same menu screen, you can select your Ring Group entry for the Additional Failover Options.

[[category:guides]]

FAQ

2013-10-16T15:33:16Z

Ed:

__TOC__

== Can I port my existing number from another provider to VoIP.ms ==
VoIP.ms does offer LNP service, and your number may be available for porting. Please contact Customer service to find out whether your current number is portable to VoIP.ms network or if you already have an open account, refer to the [[Porting a Number]] guide.

== Why am I receiving calls from an "Asterisk" caller ID? ==
If you are receiving calls with '''“Asterisk”''' as the caller ID name, the first suggestion is to check is if the calls appear on your VoIP.ms CDR as incoming calls. If they don't, then you can be sure that these calls are not coming through Voip.ms system since every call that reaches our system will appear on your VoIP.ms CDR.

If this is the case, one of the most probable reasons is a direct SIP call to your Asterisk System or Device, for Spam, SIP campaigns or scanners searching for public IP addresses with SIP port open in order to Brute force.
Unfortunately this is out of Voip.ms control as these calls never reach our system.

However, there are some suggestions you could follow in order to eliminate this inconvenience:

* If you are using some kind of '''Asterisk based system''', please make sure you disable availability to receive “Anonymous calls” and install proper security.

* If you are using an '''ATA, softphone or IP phone''', check for any option to disable these anonymous calls, if there is not, most likely your device doesn't support “Anonymous calls” by default.

* Additionally is recommended to verify your router is '''not set on DMZ mode''' for your ATA device´s internal IP, to make sure your devices and Network are not reachable from public Internet and avoid any unnecessary port forwarding.

* If there is an option on the ATA device to '''restrict incoming calls to only the IP address you are registered with''', switch it on. (This will help to prevent calls from other than voip.ms being accepted)

== Does VoIP.ms offer Calling Card solution to purchase credits for outgoing calls? ==
Voip.ms does not offer a calling card solution. However, you can use your DIDs and forward them to a calling card provider if you desire.
Voip.ms also offers the [[DISA]] feature (looks a little bit like the calling card feature). Where you call the DID, the system then will ask for a pin number, after entering the caller will be granted with dial tone to make the final call via VoIP while paying the VoIP.ms rates when calling to anywhere in the world.

== Where is VoIP.ms located ? ==

Voip.ms is a Canadian Company. The main office is located in Montreal, Canada and there is also an office in Merida, Mexico for their South America market. Credit Card charges are made in US Dollars (USD) Currency by Swiftvox Inc via Internet secure.

== What are the IP addresses of VoIP.ms´ servers ? ==

*Atlanta, GA ('''atlanta.voip.ms''') 63.247.78.218
*Atlanta 2, GA ('''atlanta2.voip.ms''') 72.9.246.170
*Chicago, IL ('''chicago.voip.ms''') 208.100.39.52
*Chicago 2, IL ('''chicago2.voip.ms''') 208.100.39.53
*Chicago 3, IL ('''chicago3.voip.ms''') 208.100.39.54
*Chicago 4, IL ('''chicago4.voip.ms''') 208.100.39.55
*Dallas, TX ('''dallas.voip.ms''') 74.54.54.178
*Denver 1, CO ('''denver.voip.ms''') 173.248.161.90
*Denver 2, CO ('''denver2.voip.ms''') 173.248.159.210
*Houston, TX ('''houston.voip.ms''') 209.62.1.2
*Los Angeles, CA ('''losangeles.voip.ms''') 96.44.149.186
*Los Angeles 2, CA ('''losangeles2.voip.ms''') 96.44.149.202
*New York, NY ('''newyork.voip.ms''') 74.63.41.218
*New York 2, NY ('''newyork2.voip.ms''') 107.6.67.236
*New York 3, NY ('''newyork3.voip.ms''') 107.6.67.237
*New York 4, NY ('''newyork4.voip.ms''') 107.6.67.238
*Seattle, WA ('''seattle.voip.ms''') 50.23.160.50
*Seattle 2, WA ('''seattle2.voip.ms''') 50.23.160.51
*Seattle 3, WA ('''seattle3.voip.ms''') 50.23.160.52
*Tampa, FL ('''tampa.voip.ms''') 68.233.226.97
*Washington, DC ('''washington.voip.ms''') 208.43.234.226
*Washington 2, DC ('''washington2.voip.ms''') 208.43.234.227
*Montreal,QC ('''montreal.voip.ms''') 67.205.74.184
*Montreal 2,QC ('''montreal2.voip.ms''') 67.205.74.187
*Montreal 3, QC ('''montreal3.voip.ms''') 72.55.168.18
*Montreal 4, QC ('''montreal4.voip.ms''') 67.205.74.179
*Toronto 2, ON ('''toronto2.voip.ms''') 184.75.215.114
*Toronto 3, ON ('''toronto3.voip.ms''') 184.75.215.146
*Toronto 4, ON ('''toronto4.voip.ms''') 184.75.213.210
*Toronto, ON ('''toronto.voip.ms''') 184.75.215.106
*London, UK ('''london.voip.ms''') 5.77.36.136

== What server should I use ? ==

Usually in order to get better results, you should choose the server closest to your location. You can still send a ping to any of the servers to check the best response time.

== Do you offer Conference calls ? ==

Voip.ms doesn't offer Conference calls as a feature, however if your system or device is capable of establishing a conference call, then you will have no issues running a conference using VoIP.ms

== Does VoIP.ms offer a STUN server ? ==

Voip.ms doesn't provide any kind of STUN server nor is it required to use one with their service. If for any specific reason you wish to do so, you can still find a public STUN server on the web for free if you need one.

== Does VoIP.ms encrypt the communication ? ==

The SIP communication is secure although not encrypted. However, the passwords are MD5 hashed and are not transmitted without encryption when establishing the call.

== Does VoIP.ms provide a Hardware device to use the service ? ==

VoIP.ms does not provide any kind of hardware device, software or system to use the service. The service is a BYOD (Bring your own device).
You should be able to get one from any Communications specialized store, and all SIP-compatible devices are supported. (Almost all VoIP devices support SIP protocol)

== Can I use my existing device with VoIP.ms ? ==

Basically, any device or system which supports SIP or IAX2 protocol will work with Voip.ms system. If you bring your device(ATA, IP phone) from a previous provider, make sure its not locked and you are able to make changes to its configuration.
Feel free to ask Technical support for further details, however they can not help you regarding how to unlock a device from other provider (for legal reasons).

== Can I call an International Toll Free number ? ==

At this moment calls to International Toll Free numbers are not supported. Only to US and Canada toll free numbers.

== Does VoIP.ms offer alternative ports besides 5060 ? ==

Voip.ms offers alternative SIP ports, UDP 5080 and 42872 on all of their servers, You can try those ports in case your ISP does block the SIP PORT 5060 UDP or if you need to use another one.

== Does VoIP.ms offer a referral program ? ==

Voip.ms is not offering any kind of referral program at the moment.

== Can I register 2 different devices with the same account ? ==

This is strongly not suggested, it can cause conflicts while routing the calls and can steal each other device's registration.
If you need to register more than one device, please create and use [[Sub Accounts]], you will get new credentials for any additional device.

== How do I port out my number ? ==

If you have a DID number with VoIP.ms and wish to port it out to another provider, you need to always start this process on the new provider's side. They should be able to provide all the information required to start this process.

== How do I change the US48/Canada termination route for my subaccount ? ==

The US48/Canada termination route for the sub-accounts needs to be changed from the Main Menu >> [[Account Settings]] >> Account Routing tab. The route you use for the Main account is what will apply to the whole account, unlike the International route, which is independent per accounts/[[Sub Accounts]].

== Does VoIP.ms offer Distinctive Ring tones ? ==

Distinctive ring tones are not available at the moment, if you are in a situation where you need to know which one of your DIDs is receiving a call, a good option is for you to use the "Caller ID name Prefix" for the DID, to add a specific prefix to DID, different from the others.

== Can I deposit less than $25 USD ? ==

The minimum amount to deposit currently is $25 USD. However, if at anytime you don't feel satisfied with the service provided, you can ask for a refund of the remaining balance on your account.

== What is my Main Account SIP password ? ==

The main account SIP password by default is the same as your customer portal password. If you have not changed the SIP password, it is the same password you use to log into your portal. To change this password please refer to our Account Settings page: [[Account_Settings#Security|Settings Security]]

== Does VoIP.ms offer Call transfers ? ==

VoIP.ms does offer Call transfers, this needs to be requested to the VoIP.ms staff, it is not a setting you can enable from the customer portal.

== Does VoIP.ms offer wholesale rates ? ==

If you are interested on a discount based on Traffic usage or volume, please send an email to sales@voip.ms providing all details (destinations you need to call, average of minutes used per month), in order to receive proper information.

== Does VoIP.ms apply taxes ? ==

All customers with Canada as country will pay for the tax called GST. The customers who besides Canada are also from Quebec they pay an additional province tax, PST. This applies for both Paypal and CC payments. The HST will get applied on Canadian provinces that use it.

In order to get further details about how these are applied, feel free to check this link: http://en.wikipedia.org/wiki/Sales_taxes_in_Canada

[[Category: Guides]]

Cisco SPA2102 Phone Adapter with Router

2013-08-30T20:42:34Z

Ed:

[[File:Spa2102b.jpg|300px|thumb|left|Cisco SPA2102 Phone Adapter with Router]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Configuration Details==

''This is a guide for the Initial Configuration of the SPA2102/3102.''

'''Step 1'''

*Power off your network devices, including your modem and PC.
*Connect an Ethernet Cable from the Ethernet port of the SPA to the Ethernet port of the PC.
*Connect an Ethernet Cable from the Internet port of the SPA to the LAN/Ethernet port of the Modem.
*Connect your regular handset phone, to the Line port of the SPA2102.
*Then power up modem, then the SPA2102 and then the PC.
*Launch a web browser from the PC and enter "'''http://192.168.0.1/advanced'''" in the URL address bar field.

After these steps you should now have access to the Web Interface page of the SPA2102 to start with the initial configuration.

'''NOTE: If this is not working or the Browser can't find the page, you may also need to enable the Administration web service.<br>'''
'''Dial 7932#, then when prompted press 1 to enable and 1 to confirm'''<br>'''

You can also check this information from the Linksys Quick Guide: http://wiki.voip.ms/files/linksys-spa-2102-user-guide.pdf

'''Step 2'''

You should now see the web interface of your Linksys/Sipura.

'''click on the link "Admin", and once the page has reloaded, click again on the link "Advanced View".'''

'''Step 3'''

'''Under the LINE 1 Tab, Find the following fields and fill them with the following information'''

'''Nat Keep Alive:''' Yes<br>
'''Nat Mapping/Traversal:''' Yes<br>
'''Proxy:''' atlanta.voip.ms (You can choose any of our multiple VoIP.ms servers)<br>
'''Register Expires:''' 180<br>
'''Proxy Fallback Intvl:''' 180<br>
'''Display Name:''' John Smith (Replace with your name or company name)<br>
'''User ID:''' 100000 (Replace with your VoIP.ms username)<br>
'''Password:''' ******** (Type in the account password)<br>
'''Use DNS SRV:''' NO<br>
'''DNS SRV Auto Prefix:''' NO

'''Step 3.5'''

'''On the SIP tab, Under NAT Support Parameters, set the following.'''

'''Handle VIA received:''' Yes<br>
'''Substitute VIA Addr:''' Yes<br>

'''Step 4 (Optional)'''

Optionally, you can configure your adapter with a better dial plan, allowing faster dialing of 10 digits number (Local US/Canada) and also enable 7 digits dialing in one area code of your choice.

'''At the bottom of Line 1 TAB, you will find a field called Dial Plan'''

Replace the 555 digits in the following line by the area code of your choice and copy the line, including parenthesis, in the Dialplan field in Line 1 Tab at the bottom of the page:

(911S0|310xxxx|<:1555>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|*xx.|[3468]11|822|0|00|[2-9]xxxxxx|4xxx|**275x.|xxxxxxxxxxxx.)

'''Step 5'''

Click on the "Save Settings " button at the bottom of the form.

<div style="width:100%;overflow:hidden;clear:both"></div>

== How to avoid the long delay to hear the ringtone ==

If you ever experience some delay to hear the ringtone when you make outgoing calls with your SPA. Changing the SPA's Interdigit Long Timer value can help resolve the issue. Follow the next steps in order to change that setting:

'''Note''': However before changing that option, test if calling the number with an # at the end of the number works(e.g. 5554441234#).
If that doesn't work you need to contact the support staff in voip.ms.

*1- First access the SPA's web interface.
*2- Click on the '''Admin Login''' and then click on the '''Voice''' tab.
*3- Click on the '''Regional''' tab and look for the '''Control Timer Values (sec)''' section.
*4- Enter the desire value in the '''Interdigit Long Timer''' field (for example lower this value to 4).

[[File:Ctrl timer values.jpg|800px]]

'''Note''': The image correspond to the PAP2 device. If your device doesn't have this setting, contact voip.ms support.

'''Additionally, whenever the registration is dropped and its having problems to gain it back, you can try this simple trick:'''
'''On Advanced view, go to the Line tab, look for the SIP port setting, and change it to another, i.e. if you have port 5060 change to 5080, if it is 5061 you can change to 5081.'''

== Known Issues ==

If you are receiving weird phone calls that do not show up in your CDR then these SIP Scanners trying to use your system.

Please Login to your Cisco Adapter and from Admin Login > Advanced > Voice > System > System Configuration > Restricted Access Domain

In this field put the servers you connect to: I.E. montreal.voip.ms, toronto.voip.ms

Once the Restricted Access Domain field is populated with Point-of-Presence (POP)servers, any server specified in Proxy field (Voice > LINE1> Proxy & Registration)

will not be able to establish registration UNLESS the same server is included in the Restricted Access Domain field

== See also ==
* [[Dial Plan for Linksys ATAs]]

[[Category:Analog Telephone Adapters]]

[http://tinyurl.com/lf7f3bo PDF Owner`s Manual]

Cisco SPA2102 Phone Adapter with Router

2013-08-30T20:37:46Z

Ed:

[[File:Spa2102b.jpg|300px|thumb|left|Cisco SPA2102 Phone Adapter with Router]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Configuration Details==

''This is a guide for the Initial Configuration of the SPA2102.''

'''Step 1'''

*Power off your network devices, including your modem and PC.
*Connect an Ethernet Cable from the Ethernet port of the SPA to the Ethernet port of the PC.
*Connect an Ethernet Cable from the Internet port of the SPA to the LAN/Ethernet port of the Modem.
*Connect your regular handset phone, to the Line port of the SPA2102.
*Then power up modem, then the SPA2102 and then the PC.
*Launch a web browser from the PC and enter "'''http://192.168.0.1/advanced'''" in the URL address bar field.

After these steps you should now have access to the Web Interface page of the SPA2102 to start with the initial configuration.

'''NOTE: If this is not working or the Browser can't find the page, you may also need to enable the Administration web service.<br>'''
'''Dial 7932#, then when prompted press 1 to enable and 1 to confirm'''<br>'''

You can also check this information from the Linksys Quick Guide: http://wiki.voip.ms/files/linksys-spa-2102-user-guide.pdf

'''Step 2'''

You should now see the web interface of your Linksys/Sipura.

'''click on the link "Admin", and once the page has reloaded, click again on the link "Advanced View".'''

'''Step 3'''

'''Under the LINE 1 Tab, Find the following fields and fill them with the following information'''

'''Nat Keep Alive:''' Yes<br>
'''Nat Mapping/Traversal:''' Yes<br>
'''Proxy:''' atlanta.voip.ms (You can choose any of our multiple VoIP.ms servers)<br>
'''Register Expires:''' 180<br>
'''Proxy Fallback Intvl:''' 180<br>
'''Display Name:''' John Smith (Replace with your name or company name)<br>
'''User ID:''' 100000 (Replace with your VoIP.ms username)<br>
'''Password:''' ******** (Type in the account password)<br>
'''Use DNS SRV:''' NO<br>
'''DNS SRV Auto Prefix:''' NO

'''Step 3.5'''

'''On the SIP tab, Under NAT Support Parameters, set the following.'''

'''Handle VIA received:''' Yes<br>
'''Substitute VIA Addr:''' Yes<br>

'''Step 4 (Optional)'''

Optionally, you can configure your adapter with a better dial plan, allowing faster dialing of 10 digits number (Local US/Canada) and also enable 7 digits dialing in one area code of your choice.

'''At the bottom of Line 1 TAB, you will find a field called Dial Plan'''

Replace the 555 digits in the following line by the area code of your choice and copy the line, including parenthesis, in the Dialplan field in Line 1 Tab at the bottom of the page:

(911S0|310xxxx|<:1555>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|*xx.|[3468]11|822|0|00|[2-9]xxxxxx|4xxx|**275x.|xxxxxxxxxxxx.)

'''Step 5'''

Click on the "Save Settings " button at the bottom of the form.

<div style="width:100%;overflow:hidden;clear:both"></div>

== How to avoid the long delay to hear the ringtone ==

If you ever experience some delay to hear the ringtone when you make outgoing calls with your SPA. Changing the SPA's Interdigit Long Timer value can help resolve the issue. Follow the next steps in order to change that setting:

'''Note''': However before changing that option, test if calling the number with an # at the end of the number works(e.g. 5554441234#).
If that doesn't work you need to contact the support staff in voip.ms.

*1- First access the SPA's web interface.
*2- Click on the '''Admin Login''' and then click on the '''Voice''' tab.
*3- Click on the '''Regional''' tab and look for the '''Control Timer Values (sec)''' section.
*4- Enter the desire value in the '''Interdigit Long Timer''' field (for example lower this value to 4).

[[File:Ctrl timer values.jpg|800px]]

'''Note''': The image correspond to the PAP2 device. If your device doesn't have this setting, contact voip.ms support.

'''Additionally, whenever the registration is dropped and its having problems to gain it back, you can try this simple trick:'''
'''On Advanced view, go to the Line tab, look for the SIP port setting, and change it to another, i.e. if you have port 5060 change to 5080, if it is 5061 you can change to 5081.'''

== Known Issues ==

If you are receiving weird phone calls that do not show up in your CDR then these SIP Scanners trying to use your system.

Please Login to your Cisco Adapter and from Admin Login > Advanced > Voice > System > System Configuration > Restricted Access Domain

In this field put the servers you connect to: I.E. montreal.voip.ms, toronto.voip.ms

Once the Restricted Access Domain field is populated with Point-of-Presence (POP)servers, any server specified in Proxy field (Voice > LINE1> Proxy & Registration)

will not be able to establish registration UNLESS the same server is included in the Restricted Access Domain field

== See also ==
* [[Dial Plan for Linksys ATAs]]

[[Category:Analog Telephone Adapters]]

[http://tinyurl.com/lf7f3bo PDF Owner`s Manual]

Choosing Server

2013-08-02T15:38:55Z

Ed:

[[File:Serverlocation3.png]]

= Choosing a Server =

[http://www.voip.ms VoIP.ms] offers 13 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. 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 "ping", which will provide you the latency between you and the server. Therefore the server which provides you less latency should be used.

Atlanta, GA '''atlanta.voip.ms'''
Atlanta 2, GA '''atlanta2.voip.ms'''
Chicago, IL '''chicago.voip.ms'''
Chicago 2, IL '''chicago2.voip.ms'''
Chicago 3, IL '''chicago3.voip.ms'''
Chicago 4, IL '''chicago4.voip.ms'''
Dallas, TX '''dallas.voip.ms'''
Denver 1, CO '''denver.voip.ms'''
Denver 2, CO '''denver2.voip.ms'''
Houston, TX '''houston.voip.ms'''
Los Angeles, CA '''losangeles.voip.ms'''
Los Angeles 2, CA '''losangeles2.voip.ms'''
New York, NY '''newyork.voip.ms '''
New York 2, NY '''newyork2.voip.ms'''
New York 3, NY '''newyork3.voip.ms'''
New York 4, NY '''newyork4.voip.ms'''
Seattle, WA '''seattle.voip.ms '''
Seattle 2, WA '''seattle2.voip.ms'''
Seattle 3, WA '''seattle3.voip.ms'''
Tampa, FL '''tampa.voip.ms'''
Montreal,QC '''montreal.voip.ms'''
Montreal 2,QC '''montreal2.voip.ms'''
Montreal 3, QC '''montreal3.voip.ms'''
Montreal 4, QC '''montreal4.voip.ms'''
Toronto 2, ON '''toronto2.voip.ms'''
Toronto, ON '''toronto.voip.ms'''
London, UK '''london.voip.ms'''

= What is a Ping? =

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).

Ping tools are part of Windows, Mac OS X and Linux as well as some routers.

== How does the ping work? ==

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).

== How to send a ping? ==

*Windows: Open up the DOS prompt and enter cmd /k ping xxxx.voip.ms on the box in windows (replace xxxx for the name of the server you want to ping. e.g: atlanta.voip.ms, montreal.voip.ms, etc).

*Mac: Open a "Console Application" and write "ping xxxx.voip.ms" (replace xxxx for the name of the server you want to ping). To stop pinging the server and get summary statistics, press control-c.

''If pings results are not consistent, you may have an issue with Jitter. You can work on this issue by adjusting the "Network Jitter Level" 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.''

The following is the output of running ping with the target losangeles.voip.ms.

#ping losangeles.voip.ms
Ping to losangeles.voip.ms [67.215.241.250] with 32 bytes de datos:
Response from 67.215.241.250: bytes=32 time=67ms TTL=52
Response from 67.215.241.250: bytes=32 time=69ms TTL=52
Response from 67.215.241.250: bytes=32 time=68ms TTL=52
Response from 67.215.241.250: bytes=32 time=67ms TTL=52
ping statistics from 67.215.241.250:
4 packets transmitted, 4 received, 0% packet lost. rtt min/avg/max/mdev = 67ms, 69ms, 67ms

* Sample Linux shell script to ping several voip.ms servers

#!/bin/sh
# Ping several servers and display Latency, Jitter and Packet Loss
#
# First, create a text file with all servers you want to ping - one host name per line.
# The list of voip.ms servers is available at http://wiki.voip.ms/article/Choosing_Server
myHF="voip_ping_hosts.txt"
# Sample file:
# toronto.voip.ms
# montreal.voip.ms
# seattle.voip.ms
# chicago.voip.ms
# newyork.voip.ms
#
echo "============================================"
printf "%-20s %7s %8s %6s\n" "VoIP Server" "Latency" "Jitter" "Loss"
echo "============================================"
cat ${myHF} |\
while read myLn
do
ping -c 3 -w 5 -q $myLn |\
awk '/^PING / {myH=$2}
/packet loss/ {myPL=$6}
/min\/avg\/max/ {
split($4,myS,"/")
printf( "%-20s %3.1f %1.3f %4s\n", myH, myS[2], myS[4], myPL)
}'
done
echo "============================================"

Output:

============================================
VoIP Server Latency Jitter Loss
============================================
toronto.voip.ms 68.3 0.439 0%
montreal.voip.ms 89.6 0.197 0%
seattle.voip.ms 71.2 0.387 0%
chicago.voip.ms 71.6 0.084 0%
newyork.voip.ms 79.1 0.411 0%
============================================

= Latency and its importance =

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.

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.

[[category:guides]]

Configuración de DID (Manage DID)

2013-08-02T15:20:28Z

Ed:

En este artículo se tratará de explicar de manera breve y básica la forma en la que usted puede configurar su propio número DID.

Una vez que usted ya ha ordenado un DID y lo tiene listo en su cuenta, es momento de hacer cambios en su configuración para recibir las llamadas de la manera deseada.
Para accesar estas opciones hay que ir al menú '''DID numbers''' y seleccionar la opción '''Manage DID(s)'''.

[[File:MenuOption.jpg]]

Una vez que se selecciona la opción '''Manage DID(s)''', sigue una pantalla en la que se encuentran listados todos los DID pertenecientes a su cuenta. Igualmente usted verá varios iconos y palabras que pueden no ser tan familiares, pero puede obtener una rápida descripción al hacer click en el icono de '''"Help"''' en la parte superior derecha.

[[File:ManageDID.jpg]]

La manera más fácil para editar la configuración de un DID, es haciendo click en el pequeño ícono con el papel y el lápiz.

[[File:Icon.jpg]]

La primera y principal sección en esta pagina es '''"Routing"''', esta sección es la que define la ruta y el comportamiento que seguirán las llamadas entrantes para el DID seleccionado.
Las diferentes opciones pueden o no aparecer todas disponibles, esto depende únicamente de si usted ya las ha creado con anterioridad.
Por ejemplo, si no ha creado un [[Buz%C3%B3n_de_voz_(Voicemail)|Buzón de voz]] - Voicemail, esta opción no aparecerá disponible.
Para seleccionar una opción, se debe hacer click en la casilla circular, y después seleccionar la opción de la lista desplegable en caso de haber más de una.

[[File:DID-Routing.jpg]]

Después de esto, tenemos otras opciones adicionales para el DID.

*La primera es '''"Additional Failover options"''', al hacer click en esta, se despliegan 3 secciones mas, idénticas a la de Routing previamente vista, pero el propósito de estas opciones, es configurar la ruta cuando la llamada alcance 1 de 3 estados diferentes, entre No disponible (''Unreachable''), Ocupado (''Busy'') y Sin Repuesta (''No answer''), en lugar de solo el [[Buz%C3%B3n_de_voz_(Voicemail)|Buzón de voz]].
*Después, tenemos la opción de '''"Voicemail"''' ([[Buz%C3%B3n_de_voz_(Voicemail)|Buzón de voz]]), es en esta parte donde se asigna un [[Buz%C3%B3n_de_voz_(Voicemail)|Buzón de voz]] al DID.
*'''"DID Point of Presence"''', esta opción es muy importante ya que es aquí donde seleccionamos en que [[Elegir_servidor|servidor]] queremos ubicar nuestro DID. Es muy importante notar que esta opción debe ser la misma que el servidor que hemos configurado en nuestro dispositivo SIP, softphone o Trunk desde un PBX.

[[File:Managedidnew.jpg]]

*'''Dial Timeout in Seconds''', esta opción es básicamente la cantidad de segundos o timbres que el numero sonará, antes de enviar la llamada al [[Buz%C3%B3n_de_voz_(Voicemail)|Buzón de voz]], o cualquiera de las "Failover options".
*'''Caller ID name Lookup''', cuando esta opción es activada, el sistema buscará en las bases de datos por un nombre asignado al [[Numero_Identificador_(Caller_ID)|número]] de la persona que le habla, si se encuentra, el nombre será desplegado junto al [[Numero_Identificador_(Caller_ID)|número]] del hablante. (Ej, en sus llamadas entrantes aparecerá JOHN SMITH, 5553332211)
*'''Caller ID name Prefix''', esta opción, es simplemente para añadir una palabra cualquiera, como un prefijo al Nombre o [[Numero_Identificador_(Caller_ID)|número]] del que llama, de forma que usted vería ''<Prefijo, Nombre, Numero>'' en su identificador de llamada.
*'''Note''', esto es simplemente para uso interno, y poner una descripción al DID.

Recuerde que ningún cambio será aplicado hasta que no se de click en el botón '''"Click here to apply changes"''' , al final de la página.

== Configurar Varios DIDs ==

Hay 2 botones adicionales que aparecen desde la Pantalla de '''"Manage DID"''' cuando selecciona esta opción desde el menú. Estas son ''"Edit Selection - All settings at Once"'' and ''"Edit Selection - One Setting at a Time"''.

'''"Edit Selection - All settings at Once"''', al seleccionar esta opción, se ingresa a la página para editar DIDs, es importante notar, que en la forma en la que esta página termine configurada al final, es exactamente como serán configurados todos los DIDs seleccionados, esto aplica a las opciones que sufrieron cambios, y a las que no también.
Por ejemplo, si usted asigna un [[Buz%C3%B3n_de_voz_(Voicemail)|Buzón de voz]], este mismo será asignado a todos los DID seleccionados, y si deja algún campo en blanco, como el de '''"Note"''', será aplicado en blanco para todos los DIDs, si alguno tenia una nota, ésta se perderá.

'''"Edit Selection - One Setting at a Time"''', al seleccionar esta opción, se ingresa a la misma pagina de edición, con la diferencia que esta vez, cada una de las opciones tiene su propio e independiente botón para guardar los cambios, de esta forma puede hacer cambios a varios DIDs, para una opción especifica, sin cambiar el resto de las opciones.
Muy útil si sus DIDs ya están configurados, y una nueva opción surge para aplicarse, pero no quiere tocar el resto de las opciones.

[[category:Guías]]

Manage DID

2013-08-02T15:18:34Z

Ed:

Once you have DID numbers in your account, you will have to edit their default settings in order to receive/forward/route the incoming calls the way you need to. To do this you need to access the "'''Manage DID(s)'''" option from the "'''DID Numbers'''" menu tab.

[[File:MenuOption.jpg]]

Following there is a general screen where you will be able to see all the DID numbers in your account. Here you will see different keywords and icons that you may not be familiar with, however you can get a quick description of each one of them by clicking on the "'''Help'''" icon at the top right of the page.

[[File:ManageDID.jpg]]

Now, if you need to access the options to edit a single DID, the easiest way is to click on the little "'''Paper and pencil'''" icon.

[[File:Icon.jpg]]

The first and Main section you will see, is the "'''DID Routing'''" section, which may or may not have all the options available, this simply depends on which features you have already enabled from your portal, that is, if you have not created a [[Voicemail]], you wont be able to select this option and apply it to a DID.

This is basically the route the call will take, when someone calls to the DID. Whatever you select here, is the route the call will follow and will apply for all the calls.

In order to select an option, you need to click on the radio button, and then select the desired choice from the drop-down list (in case you have more than one option for that routing).

[[File:DID-Routing.jpg]]

Next, we see the additional and mostly optional settings for the DID.
*The first option is "'''Additional Failover options'''". If you click to display, you will see 3 more DID routing sections, just like the Main, except these are meant to be used for each one of the 3 call states, Unreachable, Busy and No answer.
This allows you to set a customized routing when the call reaches 1 of the 3 Call states, instead of just going to the [[Voicemail]].

*After, we have the '''[[Voicemail]]''' setting, here is where you need to assign a Mailbox to your DID.
*'''DID point of presence''', this setting is the server where you will locate your DID, in order to route the calls, this setting must match the same [[Choosing Server|server or proxy]] you are using on your [[Welcome#Devices|ATA device]], [[Welcome#PBX|PBX]] trunk or [[Welcome#Softphones|softphone]] registration in order to receive calls.

[[File:Managedidnew.jpg]]

*'''The Dial Time Out in seconds''', is basically the time the call will ring before it reaches one of the Failover states or [[Voicemail]].
*'''Caller ID name lookup''', when you enable this option, the system will perform a Query on the Databases, looking for a name matching the number of your caller, and will display the name on the [[Caller ID]] name section of the [[Caller ID]].
*'''Caller ID name prefix''', this setting will simply add any word you set, as a prefix to the [[Caller ID]] name you receive. Will also work even if you don't have "Caller ID name lookup" enabled or if you don't receive a [[Caller ID]] name. This option is specially useful when you need to differentiate incoming calls from different DIDs going to the same phone.
*'''Note''', this is just an Internal description for the DID, so you can manage them.

'''Remember that none of these changes will be saved until you press the "Click here to apply changes settings" button.'''

== Editing Multiple DIDs at a time ==

There are 2 additional buttons you will see when you access the Manage DID option from the menu, those are ''"Edit Selection - All settings at Once"'' and ''"Edit Selection - One Setting at a Time"''

'''"Edit Selection - All settings at Once"''', if you choose this option, you will enter an Edition page, the way the settings on this page end, is the exact way the settings will be applied to all the DIDs selected for edition. Be careful, as the final settings from the Edition page will be applied exactly as you see them, even those settings you did not change, will be replaced for the final configuration of the Edition page. That is if you select a [[Voicemail]], all the DIDs will use the same [[Voicemail]], if you leave a blank field like the "Caller ID prefix", all of the DIDs will have this setting blank.
This option is useful if you want all your DIDs with the exact same configuration for each one of the settings, including the NOTE.

'''"Edit Selection - One Setting at a Time"''', this option allows you to access the Edition page, but this time, every setting will have its own independent "Apply" button, so you can make a change to a specific setting, without affecting the rest.
This option is useful if you already have your DIDs configured with different settings, and you need to set only a specific setting for all of them. By doing it this way, the rest of the settings from the DIDs will remain untouched.

[[category:guides]]

File:Managedidnew.jpg

2013-08-02T15:17:47Z

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File:Managedidnew.jpg

2013-08-02T15:13:03Z

Ed:

FAQ

2013-08-02T15:06:26Z

Ed:

__TOC__

== Can I port my existing number from another provider to VoIP.ms ==
VoIP.ms does offer LNP service, and your number may be available for porting. Please contact Customer service to find out whether your current number is portable to VoIP.ms network or if you already have an open account, refer to the [[Porting a Number]] guide.

== Why am I receiving calls from an "Asterisk" caller ID? ==
If you are receiving calls with '''“Asterisk”''' as the caller ID name, the first suggestion is to check is if the calls appear on your VoIP.ms CDR as incoming calls. If they don't, then you can be sure that these calls are not coming through Voip.ms system, since every call that reaches their system, will appear on your VoIP.ms CDR.

If this is the case, one of the most probable reasons is a direct SIP call to your Asterisk System or Device, for Spam, SIP campaigns or scanners searching for public IP addresses with SIP port open in order to Brute force.
Unfortunately this is out of Voip.ms control as these calls never reach their system.

However, there are some suggestions you could follow in order to eliminate this inconvenience:

* If you are using some kind of '''Asterisk based system''', please make sure you disable availability to receive “Anonymous calls” and install proper security.

* If you are using an '''ATA, softphone or IP phone''', check for any option to disable these anonymous calls, if there is not, most likely your device doesn't support “Anonymous calls” by default.

* Additionally is recommended to verify your router is '''not set on DMZ mode''' for your ATA device´s internal IP, to make sure your devices and Network are not reachable from public Internet and avoid any unnecessary port forwarding.

* If there is an option on the ATA device to '''restrict incoming calls to only the IP address you are registered with''', switch it on. (This will help to prevent calls from other than voip.ms being accepted)

== Does VoIP.ms offer Calling Card solution to purchase credits for outgoing calls? ==
Voip.ms does not offer a calling card solution. However, you can use your DIDs and forward them to a calling card provider if you desire.
Voip.ms also offers the [[DISA]] feature (looks a little bit like the calling card feature). Where you call the DID, the system then will ask for a pin number, after entering the caller will be granted with dial tone to make the final call via VoIP while paying the VoIP.ms rates when calling to anywhere in the world.

== Where is VoIP.ms located ? ==

Voip.ms is a Canadian Company. The main office is located in Montreal, Canada and there is also an office in Merida, Mexico for their South America market. Credit Card charges are made in US Dollars (USD) Currency by Swiftvox Inc via Internet secure.

== What are the IP addresses of VoIP.ms´ servers ? ==

*Atlanta, GA ('''atlanta.voip.ms''') 174.34.146.162
*Atlanta 2, GA ('''atlanta2.voip.ms''') 72.9.246.170
*Chicago, IL ('''chicago.voip.ms''') 208.100.39.52
*Chicago 2, IL ('''chicago2.voip.ms''') 208.100.39.53
*Chicago 3, IL ('''chicago3.voip.ms''') 208.100.39.54
*Chicago 4, IL ('''chicago4.voip.ms''') 208.100.39.55
*Dallas, TX ('''dallas.voip.ms''') 74.54.54.178
*Denver 1, CO ('''denver.voip.ms''') 173.248.161.90
*Denver 2, CO ('''denver2.voip.ms''') 173.248.159.210
*Houston, TX ('''houston.voip.ms''') 209.62.1.2
*Los Angeles, CA ('''losangeles.voip.ms''') 67.215.241.250
*Los Angeles 2, CA ('''losangeles2.voip.ms''') 96.44.149.202
*New York, NY ('''newyork.voip.ms''') 74.63.41.218
*New York 2, NY ('''newyork2.voip.ms''') 107.6.67.236
*New York 3, NY ('''newyork3.voip.ms''') 107.6.67.237
*New York 4, NY ('''newyork4.voip.ms''') 107.6.67.238
*Seattle, WA ('''seattle.voip.ms''') 69.147.236.82
*Seattle 2, WA ('''seattle2.voip.ms''') 50.23.160.51
*Seattle 3, WA ('''seattle3.voip.ms''') 50.23.160.52
*Tampa, FL ('''tampa.voip.ms''') 68.233.226.97
*Montreal,QC ('''montreal.voip.ms''') 67.205.74.184
*Montreal 2,QC ('''montreal2.voip.ms''') 67.205.74.187
*Montreal 3, QC ('''montreal3.voip.ms''') 72.55.168.18
*Montreal 4, QC ('''montreal4.voip.ms''') 67.205.74.179
*Toronto 2, ON ('''toronto2.voip.ms''') 199.21.149.152
*Toronto, ON ('''toronto.voip.ms''') 199.21.149.36
*London, UK ('''london.voip.ms''') 5.77.36.136

== What server should I use ? ==

Usually in order to get better results, you should choose the server closest to your location. You can still send a ping to any of the servers to check the best response time.

== Do you offer Conference calls ? ==

Voip.ms doesn't offer Conference calls as a feature, however if your system or device is capable of establishing a conference call, then you will have no issues running a conference using VoIP.ms

== Does VoIP.ms offer a STUN server ? ==

Voip.ms doesn't provide any kind of STUN server nor is it required to use one with their service. If for any specific reason you wish to do so, you can still find a public STUN server on the web for free if you need one.

== Does VoIP.ms encrypt the communication ? ==

The SIP communication is secure although not encrypted. However, the passwords are MD5 hashed and are not transmitted without encryption when establishing the call.

== Does VoIP.ms provide a Hardware device to use the service ? ==

VoIP.ms does not provide any kind of hardware device, software or system to use the service. The service is a BYOD (Bring your own device).
You should be able to get one from any Communications specialized store, and all SIP-compatible devices are supported. (Almost all VoIP devices support SIP protocol)

== Can I use my existing device with VoIP.ms ? ==

Basically, any device or system which supports SIP or IAX2 protocol will work with Voip.ms system. If you bring your device(ATA, IP phone) from a previous provider, make sure its not locked and you are able to make changes to its configuration.
Feel free to ask Technical support for further details, however they can not help you regarding how to unlock a device from other provider (for legal resons).

== Can I call an International Toll Free number ? ==

At this moment calls to International Toll Free numbers are not supported. Only to US and Canada toll free numbers.

== Does VoIP.ms offer alternative ports besides 5060 ? ==

Voip.ms offers alternative SIP ports, UDP 5080 and 42872 on all of their servers, You can try those ports in case your ISP does block the SIP PORT 5060 UDP or if you need to use another one.

== Does VoIP.ms offer a referral program ? ==

Voip.ms is not offering any kind of referral program at the moment.

== Can I register 2 different devices with the same account ? ==

This is strongly not suggested, it can cause conflicts while routing the calls and can steal each other device's registration.
If you need to register more than one device, please create and use [[Sub Accounts]], you will get new credentials for any additional device.

== How do I port out my number ? ==

If you have a DID number with VoIP.ms and wish to port it out to another provider, you need to always start this process on the new provider's side. They should be able to provide all the information required to start this process.

== How do I change the US48/Canada termination route for my subaccount ? ==

The US48/Canada termination route for the sub-accounts needs to be changed from the Main Menu >> [[Account Settings]] >> Account Routing tab. The route you use for the Main account is what will apply to the whole account, unlike the International route, which is independent per accounts/[[Sub Accounts]].

== Does VoIP.ms offer Distinctive Ring tones ? ==

Distinctive ring tones are not available at the moment, if you are in a situation where you need to know which one of your DIDs is receiving a call, a good option is to you to use the "Caller ID name Prefix" for the DID, to add a specific prefix to DID, different from the others.

== Can I deposit less than $25 USD ? ==

The minimum amount to deposit currently is $25 USD. However, if at anytime you don't feel satisfied with the service provided, you can ask for a refund of the remaining balance on your account.

== What is my Main Account SIP password ? ==

The main account SIP password by default is the same as your customer portal password. If you have not changed the SIP password, it is the same password you use to log into your portal. To change this password please refer to our Account Settings page: [[Account_Settings#Security|Settings Security]]

== Does VoIP.ms offer Call transfers ? ==

VoIP.ms does offer Call transfers, this needs to be requested to the VoIP.ms staff, it is not a setting you can enable from the customer portal.

== Does VoIP.ms offer wholesale rates ? ==

If you are interested on a discount based on Traffic usage or volume, please send an email to sales@voip.ms providing all details (destinations you need to call, average of minutes used per month), in order to receive proper information.

== Does VoIP.ms apply taxes ? ==

There are GST and PST as applicable for Paypal and Credit card payments. VoIP.ms started it because it is mandatory for them to do that due to the fact they are a Canadian company. In order to get further details about how these are applied please you could check this link: http://en.wikipedia.org/wiki/Sales_taxes_in_Canada ,also notice, if you are in Quebec PST is charged too.

[[Category: Guides]]

Zoiper Mobile (App)

2013-04-19T16:46:19Z

Ed: Created page with "==Configuration Setup== 250px '''Download the Zoiper app from the Android Market or the Apple App Store''' *'''Start the application''' -'''Cl..."

==Configuration Setup==

[[File:Zoiperapphome.jpg|250px]]

'''Download the Zoiper app from the Android Market or the Apple App Store'''

*'''Start the application'''
-'''Click on Config (Android) or Settings (IOS)'''
-'''Accounts'''
-'''+ Add account'''

[[File:Zoiperapp 1.jpg|400px]]

*'''Click on the + Add account sign on the top'''
-'''Click on SIP or IAX''' depending the protocol you would like to use.

'''Fill in according to your account
'''

'''Account Name:''' John Smith (your name)

'''Host:''' atlanta.voip.ms (one of our multiple servers)

'''Username:''' 100000 (your VoIP.ms username)

'''Password:''' ******** (account password)

[[File:Zoiperapp3.png|250px]]

Hit the '''Save''' Button

*You should be ready to make calls

File:Zoiperapp3.png

2013-04-19T16:42:54Z

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File:Zoiperapphome.jpg

2013-04-19T16:42:14Z

Ed:

File:Zoiperapp 1.jpg

2013-04-19T16:41:18Z

Ed:

File:Aureswald 5010 2.png

2013-04-02T14:49:02Z

Ed: uploaded a new version of "File:Aureswald 5010 2.png"

Auerswald COMpact 5010

2013-04-02T14:41:55Z

Ed:

[[File:Auerswald_5010.jpg|300px]]

After you have logged in to your device configuration screen, please enter the following information.

==Configuration Details==

'''Step 1'''

Under '''''Voice Over IP (VoIP) >> Providers''''', enter the following:

'''Domain:''' london.voip.ms (Replace with the address of one of the multiple servers from VoIP.ms)

'''Registrar:''' london.voip.ms (Replace with the address of one of the multiple servers from VoIP.ms)

'''Port:''' 5060

'''Note:'''

* The (internal) SIP port set by the box, each configuration for a provider has it's own port
* The shown codec setting works only if G.711 and G.729 are selected in the portal, otherwise you have to select only these two codecs in the PBX configuration
* Format der eigenen Rufnummer (own number): With this setting, the PBX will send the correct caller ID.

[[File:Auerswald 5010 1.jpg|1000px]]

'''Step 2'''

Under the next tab '''''Voice Over IP (VoIP) >> Accounts''''', enter the following:

Select the provider that you have entered on Step 1 (Voipms), then under Kind of Connection select PTMP Connection, finally click the button Configure.

[[File:Aureswald 5010 2.png|1000px]]

Now on the following screen, set the following.

'''Account Name:''' The name that was chosen in Setup Account 1

'''Country Prefix:''' E.g. 001 for NAMP or 01141 for Switzerland

'''Area code:''' 201 for New York

'''Country:''' Technically the country that correspondence with the country prefix, however there is a back in this version of the firmware and only "Germany" can be chosen. It does not have any effect on the functionality

'''Exchange line access number (account number)''': The internal number in the PBX for this account. Can be used to dial-out directly and override the set up for external line preference

'''User name:''' 100000 (your VoIP.ms username)

'''Password:''' ********* (account password)

'''Authorization ID:''' Can be left blank

'''Multiple subscriber numbers (MSNs):''' The phone number(s) chosen/assigned by VoIP.ms

'''Display name:''' Assign a unique display name (16 characters max.) for the MSN

'''Ringer rhythm:''' Optionally, you can select the ringer rhythm that should be used for the MS

[[File:Aureswald 5010 3.png|1000px]]

Auerswald COMpact 5010

2013-04-02T14:39:45Z

Ed:

[[File:Auerswald_5010.jpg|300px]]

After you have logged in to your device configuration screen, please enter the following information.

==Configuration Details==

'''Step 1'''

Under '''''Voice Over IP (VoIP) >> Providers''''', enter the following:

'''Domain:''' london.voip.ms (Replace with the address of one of the multiple servers from VoIP.ms)

'''Registrar:''' london.voip.ms (Replace with the address of one of the multiple servers from VoIP.ms)

'''Port:''' 5060

'''Note:'''
- The (internal) SIP port set by the box, each configuration for a provider has it's own port
- The shown codec setting works only if G.711 and G.729 are selected in the portal, otherwise you have to select only these two codecs in the PBX configuration
- Format der eigenen Rufnummer (own number): With this setting, the PBX will send the correct caller ID.

[[File:Auerswald 5010 1.jpg|900px]]

'''Step 2'''

Under the next tab '''''Voice Over IP (VoIP) >> Accounts''''', enter the following:

Select the provider that you have entered on Step 1 (Voipms), then under Kind of Connection select PTMP Connection, finally click the button Configure.

[[File:Aureswald 5010 2.png|900px]]

Now on the following screen, set the following.

'''Account Name:''' The name that was chosen in Setup Account 1

'''Country Prefix:''' E.g. 001 for NAMP or 01141 for Switzerland

'''Area code:''' 201 for New York

'''Country:''' Technically the country that correspondence with the country prefix, however there is a back in this version of the firmware and only "Germany" can be chosen. It does not have any effect on the functionality

'''Exchange line access number (account number)''': The internal number in the PBX for this account. Can be used to dial-out directly and override the set up for external line preference

'''User name:''' 100000 (your VoIP.ms username)

'''Password:''' ********* (account password)

'''Authorization ID:''' Can be left blank

'''Multiple subscriber numbers (MSNs):''' The phone number(s) chosen/assigned by VoIP.ms

'''Display name:''' Assign a unique display name (16 characters max.) for the MSN

'''Ringer rhythm:''' Optionally, you can select the ringer rhythm that should be used for the MS

[[File:Aureswald 5010 3.png|900px]]

Auerswald COMpact 5010

2013-04-02T14:36:25Z

Ed:

[[File:Auerswald_5010.jpg|300px]]

After you have logged in to your device configuration screen, please enter the following information.

==Configuration Details==

'''Step 1'''

Under '''''Voice Over IP (VoIP) >> Providers''''', enter the following:

'''Domain:''' london.voip.ms (Replace with the address of one of the multiple servers from VoIP.ms)

'''Registrar:''' london.voip.ms (Replace with the address of one of the multiple servers from VoIP.ms)

'''Port:''' 5060

[[File:Auerswald 5010 1.jpg|800px]]

'''Step 2'''

Under the next tab '''''Voice Over IP (VoIP) >> Accounts''''', enter the following:

Select the provider that you have entered on Step 1 (Voipms), then under Kind of Connection select PTMP Connection, finally click the button Configure.

[[File:Aureswald 5010 2.png|800px]]

Now on the following screen, set the following.

'''Account Name:''' The name that was chosen in Setup Account 1

'''Country Prefix:''' E.g. 001 for NAMP or 01141 for Switzerland

'''Area code:''' 201 for New York

'''Country:''' Technically the country that correspondence with the country prefix, however there is a back in this version of the firmware and only "Germany" can be chosen. It does not have any effect on the functionality

'''Exchange line access number (account number)''': The internal number in the PBX for this account. Can be used to dial-out directly and override the set up for external line preference

'''User name:''' 100000 (your VoIP.ms username)

'''Password:''' ********* (account password)

'''Authorization ID:''' Can be left blank

'''Multiple subscriber numbers (MSNs):''' The phone number(s) chosen/assigned by VoIP.ms

'''Display name:''' Assign a unique display name (16 characters max.) for the MSN

'''Ringer rhythm:''' Optionally, you can select the ringer rhythm that should be used for the MS

[[File:Aureswald 5010 3.png|800px]]

Auerswald COMpact 5010

2013-04-02T14:35:51Z

Ed:

[[File:Auerswald_5010.jpg|300px]]

After you have logged in to your device configuration screen, please enter the following information.

==Configuration Details==

'''Step 1'''

Under '''''Voice Over IP (VoIP) >> Providers''''', enter the following:

'''Domain:''' london.voip.ms (Replace with the address of one of the multiple servers from VoIP.ms)

'''Registrar:''' london.voip.ms (Replace with the address of one of the multiple servers from VoIP.ms)

'''Port:''' 5060

[[File:Auerswald 5010 1.jpg|700px]]

'''Step 2'''

Under the next tab '''''Voice Over IP (VoIP) >> Accounts''''', enter the following:

Select the provider that you have entered on Step 1 (Voipms), then under Kind of Connection select PTMP Connection, finally click the button Configure.

[[File:Aureswald 5010 2.png|700px]]

Now on the following screen, set the following.

'''Account Name:''' The name that was chosen in Setup Account 1

'''Country Prefix:''' E.g. 001 for NAMP or 01141 for Switzerland

'''Area code:''' 201 for New York

'''Country:''' Technically the country that correspondence with the country prefix, however there is a back in this version of the firmware and only "Germany" can be chosen. It does not have any effect on the functionality

'''Exchange line access number (account number)''': The internal number in the PBX for this account. Can be used to dial-out directly and override the set up for external line preference

'''User name:''' 100000 (your VoIP.ms username)

'''Password:''' ********* (account password)

'''Authorization ID:''' Can be left blank

'''Multiple subscriber numbers (MSNs):''' The phone number(s) chosen/assigned by VoIP.ms

'''Display name:''' Assign a unique display name (16 characters max.) for the MSN

'''Ringer rhythm:''' Optionally, you can select the ringer rhythm that should be used for the MS

[[File:Aureswald 5010 3.png|700px]]

Auerswald COMpact 5010

2013-04-02T14:33:09Z

Ed:

[[File:Auerswald_5010.jpg|300px]]

After you have logged in to your device configuration screen, please enter the following information.

==Configuration Details==

'''Step 1'''

Under ''Voice Over IP (VoIP) >> Providers'', enter the following:

'''Domain:''' london.voip.ms (Replace with the address of one of the multiple servers from VoIP.ms)

'''Registrar:''' london.voip.ms (Replace with the address of one of the multiple servers from VoIP.ms)

'''Port:''' 5060

[[File:Auerswald 5010 1.jpg|300px]]

'''Step 2'''

Under the next tab ''Voice Over IP (VoIP) >> Accounts'', enter the following:

Select the provider that you have entered on Step 1 (Voipms), then under Kind of Connection select PTMP Connection, finally click the button Configure.

[[File:Aureswald 5010 2.png|300px]]

Now on the following screen, set the following.

'''Account Name:''' The name that was chosen in Setup Account 1

'''Country Prefix:''' E.g. 001 for NAMP or 01141 for Switzerland

'''Area code:''' 201 for New York

'''Country:''' Technically the country that correspondence with the country prefix, however there is a back in this version of the firmware and only "Germany" can be chosen. It does not have any effect on the functionality

'''Exchange line access number (account number)''': The internal number in the PBX for this account. Can be used to dial-out directly and override the set up for external line preference

'''User name:''' 100000 (your VoIP.ms username)

'''Password:''' ********* (account password)

'''Authorization ID:''' Can be left blank

'''Multiple subscriber numbers (MSNs):''' The phone number(s) chosen/assigned by VoIP.ms

'''Display name:''' Assign a unique display name (16 characters max.) for the MSN

'''Ringer rhythm:''' Optionally, you can select the ringer rhythm that should be used for the MS

[[File:Aureswald 5010 3.png|300px]]

File:Aureswald 5010 3.png

2013-04-02T14:22:52Z

Ed:

File:Aureswald 5010 2.png

2013-04-02T14:22:30Z

Ed:

File:Auerswald 5010 1.jpg

2013-04-01T20:55:48Z

Ed:

Auerswald COMpact 5010

2013-04-01T20:47:01Z

Ed:

[[File:Auerswald_5010.jpg|300px]]

After you have logged in to your device configuration screen, please enter the following information.

==Configuration Details==

Step 1

Under Voice Over IP (VoIP) >> Providers, enter the following:

Domain:

Registrar:

NAT Transversal:

Interval for NAT Keep-Alive (35-255 Sec.)

Outbound Proxy

SIP Port:

Step 2

Under the next tab Voice Over IP (VoIP) >> Accounts, enter the following:

Select the provider that you have entered on Step 1 (Voipms), then under Kind of Connection select PTMP Connection, finally click the button Configure. You can see these settings on the following screen shot.

Now on the following screen, set the following.

Account Name: The name that was chosen in Setup Account 1
Country Prefix: E.g. 001 for NAMP or 01141 for Switzerland
Area code: 201 for New York
Country: Technically the country that correspondence with the country prefix, however there is a back in this version of the firmware and only "Germany" can be chosen. It does not have any effect on the functionality
Exchange line access number (account number): The internal number in the PBX for this account. Can be used to dial-out directly and override the set up for external line preference
User name: User name chosen
Password: Password chosen
Authorization ID: Can be left blank
Multiple subscriber numbers (MSNs): The phone number(s) chosen/assigned by VoIP.ms
Display name: Assign a unique display name (16 characters max.) for the MSN
Ringer rhythm: Optionally, you can select the ringer rhythm that should be used for the MS

Auerswald COMpact 5010

2013-03-28T19:08:35Z

Ed: Created page with "300px After you have logged in to your device configuration screen, please enter the following information. ==Configuration Details== Step 1 Unde..."

File:Auerswald 5010.jpg

2013-03-28T18:54:07Z

Ed:

Grandstream GXP2120 IP Phone

2012-12-05T20:01:54Z

Ed:

[[File:Gxp2110.png|300px|thumb|left|Grandstream GXP2120 IP Phone]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Configuration Detail==

'''Step 1:'''

You must first determine what IP address it received. To do this, press the round menu button, then the down arrow next to it until the display shows:

[2] IP Addr

Now, press the round menu button again and you will be shown the IP address of your device (e.g. 192.168.0.100).

'''Step 2:'''

Open your web browser and go to the IP you just found: https://[DeviceIP]

'''Step 3:'''

You will be prompted to enter a password, the default is "admin", then click the "Login" button.

'''Step 4:'''

Click on '''ACCOUNT''', follow by the '''ACCOUNT 1''' (to configure your line 1). You need to modify only a few settings from the factory default.

Fill the followings fields.

* '''SIP Server:''' atlanta.voip.ms (You can enter the address of any of our multiple servers, this is just an sample with Atlanta server)
* '''SIP User ID:''' 100000 (Replace with your VoIP.ms username, it is a six digit number)
* '''Authenticate Password:''' ********* (Type your VoIP.ms account Password)
* '''SIP Registration:''' Yes
* '''Register Expiration:''' 5

'''Step 5:'''

Finally, please click the '''Update''' button and then '''Reboot''' at the bottom of the configuration screen.

[[File:Grandstream2021final1.jpg|800px|]]

Grandstream GXP2120 IP Phone

2012-12-05T20:01:29Z

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Grandstream GXP2120 IP Phone

2012-12-05T18:24:23Z

Ed:

[[File:Gxp2110.png|300px|thumb|left|Grandstream GXP2120 IP Phone]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Configuration Detail==

'''Step 1:'''

You must first determine what IP address it received. To do this, press the round menu button, then the down arrow next to it until the display shows:

[2] IP Addr

Now, press the round menu button again and you will be shown the IP address of your device (e.g. 192.168.0.100).

'''Step 2:'''

Open your web browser and go to the IP you just found: https://[DeviceIP]

'''Step 3:'''

You will be prompted to enter a password, the default is "admin", then click the "Login" button.

'''Step 4:'''

Click on '''ACCOUNT''', follow by the '''ACCOUNT 1''' (to configure your line 1). You need to modify only a few settings from the factory default.

Fill the followings fields.

* '''SIP Server:''' atlanta.voip.ms (one of our multiple servers)
* '''SIP User ID:''' 100000 (Your VoIP.ms username)
* '''Authenticate Password:''' ********* (Account Password)
* '''SIP Registration:''' Yes
* '''Register Expiration:''' 5

[[File:Grandstream2021final1.jpg|800px|]]

Grandstream GXP2120 IP Phone

2012-12-05T18:23:28Z

Ed:

[[File:Gxp2110.png|300px|thumb|left|Grandstream GXP2120 IP Phone]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Configuration Detail==

'''Step 1:'''

You must first determine what IP address it received. To do this, press the round menu button, then the down arrow next to it until the display shows:

[2] IP Addr

Now, press the round menu button again and you will be shown the IP address of your device (e.g. 192.168.0.100).

'''Step 2:'''

Open your web browser and go to the IP you just found: https://[DeviceIP]

'''Step 3:'''

You will be prompted to enter a password, the default is "admin", then click the "Login" button.

'''Step 4:'''

Click on '''ACCOUNT''', follow by the '''ACCOUNT 1''' (to configure your line 1). You need to modify only a few settings from the factory default.

Fill the followings fields.

* '''SIP Server:''' atlanta.voip.ms (one of our multiple servers)
* '''SIP User ID:''' 100000 (Your VoIP.ms username)
* '''Authenticate Password:''' ********* (Account Password)
* '''SIP Registration:''' Yes
* '''Register Expiration:''' 5

[[File:Grandstream2021final1.jpg|600px|]]

Grandstream GXP2120 IP Phone

2012-12-05T18:23:06Z

Ed: Created page with "Grandstream GXP2120 IP Phone <div style="width:100%;overflow:hidden;clear:both"></div> ==Configuration Detail== '''Step 1:''' You must fi..."

[[File:Gxp2110.png|300px|thumb|left|Grandstream GXP2120 IP Phone]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Configuration Detail==

'''Step 1:'''

You must first determine what IP address it received. To do this, press the round menu button, then the down arrow next to it until the display shows:

[2] IP Addr

Now, press the round menu button again and you will be shown the IP address of your device (e.g. 192.168.0.100).

'''Step 2:'''

Open your web browser and go to the IP you just found: https://[DeviceIP]

'''Step 3:'''

You will be prompted to enter a password, the default is "admin", then click the "Login" button.

'''Step 4:'''

Click on '''ACCOUNT''', follow by the '''ACCOUNT 1''' (to configure your line 1). You need to modify only a few settings from the factory default.

Fill the followings fields.

* '''SIP Server:''' atlanta.voip.ms (one of our multiple servers)
* '''SIP User ID:''' 100000 (Your VoIP.ms username)
* '''Authenticate Password:''' ********* (Account Password)
* '''SIP Registration:''' Yes
* '''Register Expiration:''' 5

[[File:Grandstream2021final1.jpg|300px|]]

File:Gxp2110.png

2012-12-05T18:10:47Z

Ed:

File:Grandstream2021final1.jpg

2012-12-05T18:08:55Z

Ed:

Value vs Premium

2012-08-19T15:00:52Z

Ed:

When it comes to Termination (outbound calls), VoIP.ms offers two different routes, '''Value''' and '''Premium'''. You can choose between Value and Premium route for the continental USA/Canada calls, and also for your International calls (to any country other than the continental USA and Canada)
As they are different routes, they also offer different qualities for the service and different rates.
Following, we clarify some of the differences between each route and how to switch back and forth between them in your customer portal.

__TOC__

== Canada/US48 ==
=== Value ===

Value route offers some of the greatest rates you can find to place calls to Canada and US. The outgoing [[Service Cost|call rate]] per minute is $0.0052 (that's right, half a cent) for most of Canada and $0.0105 for calls to anywhere in the continental USA.
This is a very good option if you are looking for the best wholesale rate to pay or to offer in case you are a Reseller, while keeping a very good level of quality and reliability.

=== Premium ===

The premium route to call Canada and US has a [[Service Cost|flat rate]] of $0.0125 per minute, that is for all Canada and the continental USA.
VoIP.ms uses Tier-1 termination carriers for this route. While this route might have a higher rate per minute, you will find yourself always calling with the same level of quality and reliability.

This route is strongly suggested if you are customer willing to pay a little more for assured quality on your termination calls. Still the rate is just a little above the cent.

It is also important to note, that features like [http://www.google.com.mx/search?aq=f&gcx=w&sourceid=chrome&ie=UTF-8&q=DTMF DTMF], [[DISA]], [[Callback]] and [[Caller ID]] can only be guaranteed while using Premium route. Even when value could still deliver good results at a time, Premium route will always provide a better result regarding.
Its suggested to switch to Premium route if you experience any kind of issue related to these features.

== International ==

It also offers the 2 options, value and premium.
In this case, there is no specific rate, different countries will have different [[Service Cost|termination rates]], in some cases you will find the Premium route less expensive than the Value route.

While features like [http://www.google.com.mx/search?aq=f&gcx=w&sourceid=chrome&ie=UTF-8&q=DTMF DTMF], [[DISA]], [[Callback]] and [[Caller ID]] are not guaranteed on International calls, you certainly will find better results if you test with premium route instead of value, as well, you will find usually better quality.

Prefixes for International calls are [[Dialing_Codes#International_calls|00]] and [[Dialing_Codes#International_calls|011]], however, remember you can still use [[Dialing_Codes#International_calls|033]] for Value and [[Dialing_Codes#International_calls|044]] for Premium. These codes work as an override to the setting from the portal, so you can place a single call using one route, without having to change the settings.

== Configuration ==

All new accounts are set on the Value route by default, for both US/Canada termination and International termination.
In order to select the route you will be using, you need to go to "'''Main Menu'''", then "'''[[Account Settings]]'''" option. You will find this on the first tab "'''Account Routing'''".

The USA48/Canada route can be set between Value and Premium for your main account under your "'''[[Account Settings]]'''" and can also be set per [[Sub_Accounts|sub-account]], under your [[Sub Accounts]] settings.

The International route can be set between Value and Premium for your main account and it also can be set per [[Sub_Accounts|sub-account]], in the Edit [[Sub Accounts]] settings.

[[File:Accset rout.jpg]]

== Checking Rates ==

Since there are different routes and [[Service_Cost|termination rates]], most likely you will need to know often what route is more convenient to use, while calling a specific number.
Whether is US/Canada termination or International, you can always find the specific per minute rate to a number from your portal by going to "'''Rates'''" menu, "'''Check Rates Online'''" option.

[[File:Ratesxroute.jpg]]

[[category:guides]]

Dialing Codes

2012-08-19T14:51:09Z

Ed:

[http://voip.ms/ VoIP.ms] has different dialing codes depending on the destination you want to reach, and you can use these codes from your main or sub account depending on your needs. There are two main distinctions between the routes when dialing, local and international calls, it is important to mention, calls to United States 48 (except Hawaii & Alaska) and Canada are considered as local calls. All the other destinations will be considered as International and will use the corresponding route.

== Local calls (USA48/Canada) ==

For calls to USA48 & Canada, you only need to dial the complete 10 digits number, optionally you can dial using 11 digits, adding the prefix 1 before the number. From the [[Account Settings]] >> Account Routing, it can be set a route for these calls, there is an option to use value or premium route.

*Value: USA 48 $0.0105 - Canada starting at $0.0052

*Premium: USA 48 / Canada $0.0125. ''Note (except Yukon, North West Territories & Nunavut are considered in this section but have a different rate (value at $ 0.0102 & premium at $0.0260).''

Example 1514-316-xxxx or 514-316-xxxx (do not use the dashes when dialing)

'''USA48: The contiguous United States are the 48 U.S. states on the continent of North America that are south of Canada, plus the District of Columbia. The term excludes the states of Alaska and Hawaii, and all off-shore U.S. territories and possessions, such as Puerto Rico.
'''

== International calls ==

There are different codes that can be used to dial International numbers (Outside US48 & Canada). Hawaii and Alaska are considered in this section, since any change made on the [[Account Settings]] >> Account Routing for International calls, will affect also these destinations, this means if we set to use premium route, calls to these destinations will use also premium route, but they can be dialed as local US numbers (10 or 11 digits).

Note: Some International destinations can be dialed using only the prefix 1, this applies for countries part of the NANPA. 011 & 00 are for the rest of countries.
E.g. when dialing Hawaii, Alaska, Caribbean countries an U.S. Territories.

For dialing to countries outside US & Canada, we can follow these codes:

*011 + Country Code + number: International
*00 + Country Code + number: International
*033 + 1 + Area Code + number: US48/Canada Value (override account setting)
*044 + 1 + Area Code + number: US48/Canada Premium (override account setting)
*033 + Country Code + number: International Value (override account setting)
*044 + Country Code + number: International Premium (override account setting)

(+ signs used as reference only. Do not include + signs when dialing)

Example Dialing to Mexico: 011+country code+number 011-52-9999xxxxxx (do not use the dashes when dialing)

== Special codes ==

Some special codes that you can use with the service:

=== [[Voicemail]] Access Codes ===
<nowiki>*</nowiki>97 to access directly the Mailbox associated to the account you are dialing from. (Will prompt for Password only)

<nowiki>*</nowiki>98 to access your Voicemail and choose one of your Mailbox accounts. (Will prompt for Mailbox ID and Password)

If you don't have access to our VoIP network and would like to check your Voicemail, you can simply dial your number. Once the Voicemail system answers your call, press the asterisk key (*).

=== Account Balance ===

Dial *225 (*bal): This code allow you to access your VoIP.ms Account Balance. It can be Enable or Disable on [[Sub Accounts]].

=== SIP Broker Peering ===

Dial **275 SIP Broker Peering (e.g: **275*011188888): This code can be used for outbound calls via SIP broker.

If you want to receive calls, you will need a DID number (US/Canada) and you can use the following format *9780didnumber@sipbroker.com.

You can also access this link for more information http://sipbroker.com/sipbroker/action/login

=== Echo & DTMF test ===

4443 (Echo Test): This code is used to access the echo test, with a new account this code can be dialed even without funds, it is useful to verify the quality on your line.

4747 (DTMF Test): This code is used to access the dtmf test, with a new account this code can be dialed even without funds, it is useful to verify if the dtmf is configured properly.

=== iNums ===

iNums can be ordered for free from your account, on the order did section, one iNum is available per account and calls (incoming & outgoing) are free of charge. You can get more information also at http://www.inum.net/. iNum stands for international Number, making use of the +883 global country code. iNums are reachable from a growing list of providers.

Example 011+883+iNum or 00+883+iNum: 011-883-51000134xxxx (do not use the dashes when dialing)

== Service codes for Canada ==

'''1-555-555-0911:''' Test CallerID and [[e911]] Test: to test if your caller Id is working properly you can dial this number, also this recording will let you know if your number is activated with the e911 service.

'''411:''' Directory Assistance (Must be enabled in [[Account Settings]]), when enabled users can dial the directory assistance at a cost of $0.99 per call.

'''311:''' Non-Emergency Police, Municipal and Other Governmental Services (Canadian Servers)

'''511:''' Provision of Weather and Traveler Information Services (Canadian Servers)

'''811:''' Non-Urgent Health Teletriage / telehealth Services (Canadian Servers)

'''Important Notes:'''

* The services can only be Called from Canadian servers. There are plans on adding service codes to US servers as soon as possible.
* The services rely on your [http://wiki.voip.ms/article/Caller_ID CallerID] to route your call to the proper service line, therefore, some services may not be supported in your province or for your specific area.
* The are no charges applied to your account when you call 311, 511 or 811.

=== Canadian 2-1-1 and 3-1-1 local service numbers ===

2-1-1 (community info) is directed to various numbers, depending on your local area code or municipality. The numbers listed are from [http://www.cnac.ca/other_codes/n11/plans/.../Vancouver_211.pdf BC],
[http://www.cnac.ca/other_codes/n11/plans/documents/Ontario_211.pdf Ontario] and [http://www.cnac.ca/other_codes/n11/plans/documents/QuebecCity_211.pdf Québec] as shown by the Canadian numbering administrator, cnac.ca

Ontario provides 2-1-1 community info province-wide; a similar plan was proposed for BC but currently appears to be abandoned.

In most cases, Ontario's 2-1-1 providers cover one area code and its overlays; +1-905 (Oshawa-Hamilton-Niagara) is a notable exception as it is split among multiple local providers in the 416/905 area.

As the actual number reached by dialling 2-1-1 is area code or county-specific, it may be best to use SIP dial plan rules to map 2-1-1 to the appropriate number for your area; for instance, Toronto/Oshawa 2-1-1 (in Linksys/Cisco format dial plan) would be |<211:14163974636>S0|

{|
|-
!Area code
!2-1-1 number
!Description
|-
||+1-416 Toronto
||+1-416‐397‐4636
||Toronto (FindHelp Information Services)
|-
||+1-418 Greater Québec City
||+1-418-838-5115 or +1-877-211-9933
||Lévis ([http://211quebecregions.ca Centre d’information et de référence] de la Capitale-Nationale et de la Chaudière-Appalaches)
|-
||+1-418 MRC Haute-Yamaska
||+1-418-838-0399 or +1-877-909-0399
||Lévis (Centre d’information et de référence de la Capitale-Nationale et de la Chaudière-Appalaches)
|-
||+1-519 (all points)
||+1-519‐258‐0247 or +1‐866‐686‐0045
||Windsor (211 Windsor/Essex)
|-
||+1-604 Vancouver
||+1-604-875-6431
||Metro Vancouver ([http://www.bc211.ca Information Services Vancouver])
|-
||+1-613 (all points)
||+1-613‐761‐9076
||Ottawa (Information Ottawa)
|-
||+1-705 (except Nipissing, Sudbury)
||+1-705‐445‐0641 or +1‐866‐743‐7818
||Collingwood (Communication Connections)
|-
||+1-705 Nipissing, Sudbury and points north
||+1-807‐626‐9626
||Thunder Bay (Lakehead Social Planning Council)
|-
||+1-807 (all points)
||+1-807‐626‐9626
||Thunder Bay (Lakehead Social Planning Council)
|-
||+1-905 Halton, Wellington, Waterloo
||+1-905-825-9135 or +1-866-580-3828
||Oakville (Halton Region)
|-
||+1-905 Niagara, Brant, Haldimand
||+1-905‐682‐6611 or +1‐800‐263‐3695
||St Catharines (Information Niagara)
|-
||+1-905 Oshawa, York Region
||+1-416‐397‐4636
||Toronto (FindHelp Information Services)
|-
||+1-905 Peel, Dufferin
||+1-905‐791‐4949 or +1-866‐573‐0088
||Brampton (Region of Peel)
|}

3-1-1 is typically used in large Canadian cities (such as Ottawa and Toronto) for calls to City Hall. (In Peel Region, the number initially goes to the municipality; the user is then prompted to select a regional or municipal operator.) As such, the number at which these calls terminate will differ for each served municipality.

{|
!Area / City
!3-1-1 number
|-
||+1-604 Vancouver
||+1-604-829-4206 landline, +1-604-829-4207 wireless, +1-604-873-7000 for calls from outside Vancouver
|-
||+1-905 Brampton
||+1-905-494-7818
|-
||+1-905 Mississauga
||+1-905-615-4625
|}

=== Canada 310 service ===

In Bell Canada territory (any of area codes +1-807, 705, 519, 905, 416, 613, 819, 450, 514, 418 and their overlays), exchange 310 is a special code. These numbers are only reachable from the area where they belong, but appear to be local seven-digit calls from all exchanges in an entire area code. From the service, you will need to dial only the 7 digits to properly reach these numbers, pass a valid Canadian [http://wiki.voip.ms/article/Caller_ID CallerID] and be registered in Chicago, Seattle or any of the Canadian servers.

Example: Dial 310-xxxx (do not use the dashes when dialling), if you pass a caller id in area code 514, dialling seven digits 310-xxxx is like dialling 514-310-xxxx.

Note that, in area codes outside Canada, +1-areacode-310-xxxx is most likely just a normal number; the exchange has no special significance.
[[category:guides]]

Account Settings

2012-08-19T14:37:31Z

Ed:

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 that will affect the global settings on your account, from Device type, to routing and even security of your account.

__TOC__

== Account Routing ==

These settings define the routes the system will use when you place calls to either US, Canada, Toll Free or International Numbers.

[[File:Accset rout.jpg|800px]]

'''USA48/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.
The '''Premium''' route has a better level of quality, at price that is a little higher that the value option. The CallerID Number is guaranteed to work for this route. Please note that the '''USA48/Canada Route''' can also be set independently per sub account, under the [[Sub Accounts]] settings.

'''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.

'''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.

'''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'''.

'''Server Default:''' The Carrier for the termination of your toll-free call will be chosen depending on the geographical location of the server.
American Servers will dial through an American Carrier and Canadian servers will use a Canadian Toll-Free carrier.
'''American Carrier:''' Your Toll-Free calls will always go through an American Carrier, even if you use a Canadian server.
'''Canadian Carrier:''' Your Toll-Free calls will always go through a Canadian Carrier, even if you use an American server.

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.

'''Note''': Please refer to [[Service Cost]] for information about the rates of the outgoing calls.

== Account Restrictions ==

These settings define the restrictions the system will use when you place calls to either USA, Canada or International Numbers.

[[File:Account Restrictions.jpg|800px]]

'''Allow 411 dialing''': When enabled, you can call 411 directory assistance at the cost of $0.99 per call.

'''Note''': XXX-555-1212 is disabled within VoIP.ms. The only way to reach directory assistance is by dialing 411.

'''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.

'''Note''': This option is set to "No" by default for New accounts.

'''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.

'''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.

'''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.

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.

'''Note:''' This settings depends on the International Routing you have choose.

'''Allow Calls to Countries''': Here you can select Regions or Specific Countries to allow outgoing calls from your account.

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. If you select the Region all the countries within will be selected. You can also select all countries by checking 'Allow All'. If you call to a country not allowed in this section, the call will not be connected.

[[File:Allowed countries.jpg|800px]]

'''Note''': If you have international calls disabled, it will prevent calling to the international countries you have enabled in this list.

== General ==

These are the general settings used by the system when you make or receive calls.

[[File:General.jpg|800px]]

'''Dialing Mode''': This setting allow to set the way you're going to dial other numbers.

*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.

*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.

'''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.

'''Voicemail Associated to the Main Account''': You can select the mailbox associated to the main account.

*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 "Skip Password" option in the [[voicemail]] configuration.

*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.

'''Note:'''This setting only affects the device registered with the main account.
If you're using a [[Sub Accounts|sub account]] you need to set the '''Internal Extension Voicemail''' for that subaccount.

'''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.

== Security ==

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 "Customer Portal".

[[File:Accset sec.jpg|800px]]

'''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.
Note: You must enter your current customer portal password in order to enable this option.

'''Customer Portal Password''': Here you can change the password used to log in your VoIP.ms Customer Portal.

'''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.

'''Note:''' For default the Main SIP/IAX Password is the same you use to login in your Customer Portal.

== Inbound Settings==

In this tab, you can set the protocol used for inbound calls and the device you're using for your main account.

[[File:Inbound.jpg|800px]]

'''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.

'''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.

== Notifications ==

In this tab you can set whether to allow or not notifications in your mail when the balance in your account reach certain amount.

[[File:Accset not.jpg|800px]]

'''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).

'''Note''': Set this according to the monthly use you have in your account.

'''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 (,).

'''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.

== Default DID Routing ==

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.

[[File:Accsetdefdidrou.jpg|800px]]

'''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 "Caller ID name portion" <Caller ID number>.

'''DID POP''': Here you can set the server in which your software/device is registered to or receiving the call from.
'''Note''': Always make sure to register your software/device to the same server you set in this option to properly receive incoming calls.

[[File:Accset defdidrout2.jpg|800px]]

'''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.

'''Aditional Failover Options''': This setting let you define where to redirect the call when the destination is '''Busy''', '''Unreachable''' or '''No Answer'''.

Note: You can change these settings later or enable additional settings in your Customer Portal>>DID Numbers>>Manage DID(s)

== Newsletter ==

'''Newsletter Subscription''': You can set if you want to receive news from voip.ms

[[File:Accset news.jpg|800px]]

== Advanced ==

These options allow you to set the allowed codecs, DTMF modes and NAT options. These settings are recommended for advanced users only.

[[File:Accset adv.jpg|800px]]

'''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.

'''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.
Note: Its recommended that you select the same DTMF mode in your device.

'''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.
Note: It's recommended that you select Allow All and only change it if you have an specific reason to do so.

[[category:guides]]

Sub Accounts

2012-08-19T14:31:57Z

Ed: /* How to Create a Sub Account */

Having a Sub Account allows you to register more than one device to make or receive calls simultaneously, you can also use it as internal extensions for your office or even your house. Many of the features within voip.ms make use of the sub accounts. With this guide we are going to learn how to create and use this feature properly.

__TOC__

== How to Create a Sub Account ==

You can create as many Sub Accounts as you required and there's not any extra charge for doing this. First you need to go enter your Customer Portal and select "Create Sub Account" under the Sub Account Menu.

[[File:Newsubaccountsettings.jpg|800px]]

===Basic Options ===

These are the basic options you need to configure in order to properly create a Sub Account.

'''Protocol''': Here you can select either SIP or IAX2 protocol. This will depend on the device you want to use with this Sub Account. SIP is the recommended protocol.

'''Authentication type''': This is an option that you don't have with your main account. You can choose between '''User/Password Authentication''' or '''IP Authentication'''. Again this setting will depend on the device you're using.

-- '''User/Password Authentication''': Its the recommended setting and most devices only support this option.

-- '''IP Authentication''': Recommended for advanced users. Mostly this is used for PBX or Asterisk based servers. When you select this option you can set the IP of your device/server.

'''Note:''' When you use IP Authentication the registration status will not be displayed in your Customer Portal: Main Page.

'''Username''': This is the username of the sub account. You can use alphanumeric characters for the username (make sure your device supports the use of alphanumeric characters in the user field). The format of the username will be as follow: '''{Main Account}_{username}'''.

For example, let say that your account is ''100000'' and you set the username as ''home'' the username that you're going to use in your device will be ''100000_home''.

'''Password/IP Address''': Here you can set the password to use for this sub account. The minimum is 6 characters, although it's strongly suggested that you use more characters and also use alphanumeric characters to create a strong password.

If you're using IP Authentication here you can set the IP address of your device/server. Format e.g. 201.202.203.204.

[[File:IP AUTH.jpg|800px]]

'''Device type''': Make sure to select the correct type of device you're going to use with this sub account to properly receive incoming calls in your device.

'''CallerID Number''': Here you can set the [[Caller ID|callerID number]] if you're using an ATA device, IP Phone or Softphone. Make sure to enter a proper [[Caller ID|callerID number]] to ensure proper termination. Use only numbers in this field and the [[Caller ID|callerID number]] has to be of 10 digits long. ''Optional''

'''USA48/Canada Routing''': This define the route the system will use when you place a call to Canada or within one of the 48 inland states of the US with your Sub Account.

'''International Route''': You can select the route that is going to be use for the International Calls for this sub account.

'''Allow International Calls''': With this setting you can enable or disable the International Calls for the sub account.

'''Allow *225 for Balance''': When Enabled, calls placed to *225 will provide the Current Balance of the VoIP.ms account. When Disabled calls placed to *225 will be rejected.

'''Music on Hold''': This setting allow you to select whether or not the person calling your DID number or extension will hear Music while the call is on hold.

'''Account name or description''': This setting can help you to easy identify each sub account.

=== Advanced Options ===

These options allow you to set the allowed codecs, DTMF modes and NAT for each subaccount. These settings are recommended for advanced users only.

[[File:Subacc adv opt.jpg|800px]]

'''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.
'''Note:''' It's recommended that you select '''Allow All''' and only change it if you have an specific reason to do so.

'''DTMF Mode:''' This allows you to select the DTMF mode that is going to be used with this sub 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''.
'''Note:''' Its recommended that you select the same DTMF mode in your device.

'''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''.

=== Optional Settings ===

Although these are optional settings, you can use these settings to assign an internal extension number, voicemail and dial time out for your sub account.

[[File:Sub acc optional.jpg|800px]]

'''Internal Extension Number''': You set an internal extension number in order to call between Sub Accounts under the same Main Account. The extension number you enter in this field will have a leading 10, you can enter from 1 to 10 digits. For example if you set ''55'' the extension number will be ''1055''.
Note: Make sure that the sub accounts are registered to the same server in order to make an internal call. The call to an internal extension is FREE.

==== Sub Account as an External SIP URI ====

To use a sub account as an external [[SIP URI]], you only need to enable it as an Internal Extension first. For example, let say your Internal Extension is set to 2 (102 with the leading 10), you can be reached directly via SIP from another network with a URI that is going to look like this: ''1000002@server.voip.ms'' (Replace server.voip.ms by the server you are registered to, and the 2 by your internal extension).

'''Internal Extension VoiceMail''': Here you can select which [[voicemail]] is going to be associated with this sub account.

'''Note:''' If you associate a [[voicemail]] with this subaccount, you are going to receive a '''Message Waiting Indicator''' when you have new messages in your mailbox.
This will lead to different results depending on your type of adapter, soft phone or IP phone.
For example, when using a Linksys 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.

'''Internal Extension Ringing Time''': This is the amount of time the phone will stay ringing when you call this internal extension directly. 5 sec = 1 ring.

=== Reseller Configuration ===

Also if you're using the [[Reseller Basic Guide|Reseller Interface]], you can associate each sub account with one of your customers.

[[File:Subaccreseller.jpg|800px]]

'''Reseller Client''': Here you can select the customer that you want to associate with this subaccount. You need first to create the account of your customer using the [[Reseller Basic Guide|Reseller section]] in your Customer Portal.

'''Reseller client package''': Here you can select the package that you want to assign to your customer. You need first to create a package in the [[Reseller Basic Guide|Reseller section]].

'''next billing date''': Here you can set the next billing date. Usually the system set this automatically, but if you manually change the package associate to this subaccount, you can change the next billing date manually.

'''Charge setup fees now''': Once you check this option, the monthly fee for the package is going to be charged. You can use this option when you have applied a change in the package of the customer.

== Sub Account Reports ==

You can see the amount of minutes, number of calls and even the amount spent for each sub account. This information is useful even if you're not using the [[Reseller Basic Guide|Reseller Interface]]. You can access this from your Customer Portal>>Sub Accounts>>Sub Accounts Reports menu.

[[File:Subacc report.jpg|800px]]

'''Report Range''': You can display a report with a range of 92 days only (3 months).

'''Minutes''': The number of minutes used by this sub account in the given range. Expressed using decimal time.

'''Calls''': The number of calls made for this sub account in the given range.

'''Amount Spent''': This is the amount spent for this sub account in the given range.

== Use of the subaccount ==

Once you have created one subaccount, you can use it with most of the features available within your Customer Portal, for example with the [[Digital Receptionist (IVR)]],as an agent to receive calls from [[Calling Queues]], Routed directly to your DID numbers, as an external [[SIP URI]] to receive incoming calls from another networks, etc.

[[category:guides]]

File:Newsubaccountsettings.jpg

2012-08-19T14:15:38Z

Ed:

Panasonic KX-TGP 550

2012-04-03T14:23:00Z

Ed:

== Devices ==
* Panasonic KX-TGP-550 (cordless VoIP 'phone with one base handset, one remote handset)
* Panasonic KX-TGP-500 (cordless VoIP 'phone with one remote handset, no handset on base)
<center>
{|
||[[Image:Pana550b.jpg|300px|thumb|caption|Panasonic KX-TGP 550]]
||[[Image:Panasonic KX-TGP-500.jpg|175px|thumb|caption|Panasonic KX-TGP 500]]
|}</center>
<div style="width:100%;overflow:hidden;clear:both"></div>

==Configuration Details==

Complete the following fields:

* '''Phone number ID:''' 100000 (Enter your VoIP.ms username, it´s always a six digit number)
* '''Line ID:''' Leave this field blank.
* '''Registrar Server Address:''' atlanta.voip.ms (You can enter any of the multiple VoIP.ms servers)
* '''Registrar Server port:''' 5060 (Default SIP Port)
* '''Proxy server Address:''' atlanta.voip.ms (You can enter any of the multiple VoIP.ms servers)
* '''Proxy server port:''' 5060
* '''Presence server port:''' 5060
* '''SIP service domain:''' atlanta.voip.ms (You can enter any of the multiple VoIP.ms servers)
* '''SIP source port:''' 5060
* '''Authentication ID:''' 100000 (Enter your VoIP.ms username)
* '''Authentication password:''' ********* (VoIP.ms´s Account Password)

The default registration expiry time on this device is 3600 seconds. You may have to reduce that if you lose registration. Unfortunately, that can not be set on the web interface. You need to load a provisioning configuration file from a web or FTP server (details [http://www.voicesonic.com/panasonic/manuals/panasonic-others/TGP500-550-Admin-Guide.pdf here], page 132 for file format). The parm is REG_EXPIRE_TIME_[n]="mmm", where n is line number, and mmm is the registration interval. e.g. REG_EXPIRE_TIME_1="180" to set line 1 registration time to 180 seconds. A symptom of this problem is that the telephone web interface shows that the line is registered, but the voip.ms control panel indicates that it is not registered. This device supports multiple provisioning files in a hierarchy.

The "Phone Number" field may contain only digits, and *must* contain at least one digit, or the base unit won't try to register it. Within those restrictions it can be any number you want. When using the handset (or base unit) to select a line from which to dial, this number will be displayed next to the entry for the associated line. I think Panasonic's idea is that this is supposed to be a DID associated with the SIP address, but it doesn't have to be.

Enter your SIP username in the "Line ID" field and again in the "Authentication ID" field. If you are registering a voip.ms main account your SIP username will be your six-digit account number. If you are registering a sub-account it will be your main account number followed by an underscore followed by the sub-account number.

You can register at any voip.ms server (e.g. "atlanta.voip.ms") but you must enter your chosen server name in the "Registrar Server Address" and "Proxy Server Address" and "Service Domain" fields.

After saving these settings, check the "VoIP Status" page in the "Status" section to see if the status of this line is "registered". This product can register up to eight SIP addresses at once. If any of the SIP addresses you configured into the device do not show their status on this page as being "registered", then the status light on the base unit will flash yellow. (This is not mentioned in either the user guide nor the admin guide for this product.)

The above information applies to both the TGP-550 (where the base unit has a handset, dialpad, LCD display, etc.) and the TGP-500 (where the base unit is a faceless black box that sits in your network closet).

[[Category:SIP phones]]

Media5 Mediatrix 4100

2012-03-06T21:10:20Z

Ed:

== Configuration Details ==

*'''Step 1'''

The first step is to find out what IP Address your adapter is currently using. By default, the Mediatrix 41xx (4102S, 4104, 4108, 4116 or 4124) running DGW 2.0 gets its IP address by DHCP. If the Power LED is solid on, this means the device has an IP address. You can then go to Step2. If the Ready LED is blinking, the Mediatrix unit is looking for a DHCP server, you can then connect the unit to a network with DHCP server or do a Partial Reset.
To discover your device’s IP address, pick a phone connected on Line 1 and do the following:

'''Dial: *#*0'''

The system should now playback the IP address your device has been assigned.
(Example: 192.168.1.2)

*'''Step 2'''

'''Using your favorite web browser from a computer on the same network, point the address to the IP address of your adapter.'''
(example: http://192.168.1.2) Replace 192.168.1.2 by the IP address your device is currently using.
You should now see the web interface of your Mediatrix unit.

'''The default username is Public'''

'''The Passowrd :<Empty> (There is no password by default).'''

[[File:Mediatrix.jpg]]

*'''Step 3'''

'''Click on the SIP Menu'''

'''Sub-menu Servers'''

'''Registrar host:''' montreal.voip.ms (one of our multiple servers)

'''Proxy Host:''' montreal.voip.ms (one of our multiple servers)

Click '''Submit'''

[[File:Mediatrix2.jpg]]
*'''Step 4'''

'''Click on the SIP Menu'''

'''Sub-menu Registration'''

'''Username:''' 100000 (your VoIP.ms username)
'''Friendly Name:''' John Smith (your name or company name)
Select '''Enable''' in the Register drop-down menu

Note: Depending on the model and the number of lines, you may configure from 1 to 24 ports.

Click '''Submit'''

[[File:Mediatrix3.jpg]]

*'''Step 5'''

'''Click the SIP Menu'''

'''Sub-menu Authentication'''

Click '''Edit''' on the first row

[[File:Mediatrix4.jpg]]

'''Apply to:''' Endpoint

'''Endpoint:''' Choose the FXS port for which you want to configure the Authentication

'''Validate Realm:''' disable

'''Username:''' 1000000 (your VoIP.ms username)

'''Password:''' ******** (the account password)

Click '''Submit''' & Refresh Registration

== Optional Setting: ==

*'''Step 6'''

Click the Telephony Menu
CODECS Sub Menu
Disable G.711 a-law for both Voice and Data
Next to G.729 click Edit and change the Voice priority to 10

[[File:Mediatrix5.jpg]]

Click '''Submit'''

*'''Step 7'''

To support '''User name''' with ‘’_’’

Click the '''Call Router''' menu

Sub-menu Auto-Routing

Change the Criteria Type to '''SIP Username'''

Click '''Apply Config'''

*'''Step 8'''

If your DHCP server does not provide an SNTP Server, you can configure it manually.

Click the '''Network Menu''' and then '''sub-menu Host'''

Change the SNTP Configuration Source to Static

'''Configure the SNTP Host to:''' time.nrc.ca or any other SNTP server.

Click '''Apply Config'''

'''At the end of the configuration, if you see the message Some changes require to restart Click on services table and restart all Required Services'''

Media5 Mediatrix 4100

2012-03-06T21:09:03Z

Ed:

[[File:Mediatrix4102s.jpg]]

== Configuration Details ==

*'''Step 1'''

The first step is to find out what IP Address your adapter is currently using. By default, the Mediatrix 41xx (4102S, 4104, 4108, 4116 or 4124) running DGW 2.0 gets its IP address by DHCP. If the Power LED is solid on, this means the device has an IP address. You can then go to Step2. If the Ready LED is blinking, the Mediatrix unit is looking for a DHCP server, you can then connect the unit to a network with DHCP server or do a Partial Reset.
To discover your device’s IP address, pick a phone connected on Line 1 and do the following:

'''Dial: *#*0'''

The system should now playback the IP address your device has been assigned.
(Example: 192.168.1.2)

*'''Step 2'''

'''Using your favorite web browser from a computer on the same network, point the address to the IP address of your adapter.'''
(example: http://192.168.1.2) Replace 192.168.1.2 by the IP address your device is currently using.
You should now see the web interface of your Mediatrix unit.

'''The default username is Public'''

'''The Passowrd :<Empty> (There is no password by default).'''

[[File:Mediatrix.jpg]]

*'''Step 3'''

'''Click on the SIP Menu'''

'''Sub-menu Servers'''

'''Registrar host:''' montreal.voip.ms (one of our multiple servers)

'''Proxy Host:''' montreal.voip.ms (one of our multiple servers)

Click '''Submit'''

[[File:Mediatrix2.jpg]]
*'''Step 4'''

'''Click on the SIP Menu'''

'''Sub-menu Registration'''

'''Username:''' 100000 (your VoIP.ms username)
'''Friendly Name:''' John Smith (your name or company name)
Select '''Enable''' in the Register drop-down menu

Note: Depending on the model and the number of lines, you may configure from 1 to 24 ports.

Click '''Submit'''

[[File:Mediatrix3.jpg]]

*'''Step 5'''

'''Click the SIP Menu'''

'''Sub-menu Authentication'''

Click '''Edit''' on the first row

[[File:Mediatrix4.jpg]]

'''Apply to:''' Endpoint

'''Endpoint:''' Choose the FXS port for which you want to configure the Authentication

'''Validate Realm:''' disable

'''Username:''' 1000000 (your VoIP.ms username)

'''Password:''' ******** (the account password)

Click '''Submit''' & Refresh Registration

== Optional Setting: ==

*'''Step 6'''

Click the Telephony Menu
CODECS Sub Menu
Disable G.711 a-law for both Voice and Data
Next to G.729 click Edit and change the Voice priority to 10

[[File:Mediatrix5.jpg]]

Click '''Submit'''

*'''Step 7'''

To support '''User name''' with ‘’_’’

Click the '''Call Router''' menu

Sub-menu Auto-Routing

Change the Criteria Type to '''SIP Username'''

Click '''Apply Config'''

*'''Step 8'''

If your DHCP server does not provide an SNTP Server, you can configure it manually.

Click the '''Network Menu''' and then '''sub-menu Host'''

Change the SNTP Configuration Source to Static

'''Configure the SNTP Host to:''' time.nrc.ca or any other SNTP server.

Click '''Apply Config'''

'''At the end of the configuration, if you see the message Some changes require to restart Click on services table and restart all Required Services'''

Devices

2012-03-06T21:07:45Z

Ed:

==3COM 3108 Wireless Phone==

[[File:3com_3108_1.jpg|300px|thumb|left|3COM 3108 Wireless Phone]]

'''Product:''' 3COM 3108 Wireless Phone

'''Company:''' 3COM

'''Overview:''' The 3Com 3108 Wireless Telephone is a Session Initiation Protocol (SIP)-based wireless Voice over Internet Protocol (VoIP) telephone. SIP is an internationally recognized standard <nowiki>(IETF RFC 3261)</nowiki> for implementing VoIP. You can make and receive VoIP calls as long as your Wireless Telephone is registered with a SIP proxy server and you are operating it within range of an IEEE 802.11b/g enabled wireless network (WLAN). The SIP proxy server can belong to a wireless Internet Telephony Service Provider (ITSP) or corporate VoIP PBX system, such as the 3Com NBX® System.

[[3COM_3108_Wireless_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Aastra 6730i/6731i VoIP Phone==

[[File:Aastra6730i.jpg|300px|frame|left|Aastra 6730i VoIP Phone]]

'''Product:''' Aastra 6730i/6731i VoIP Phone

'''Company:''' Aastra

'''Overview:'''The Aastra 6730i, a new member of the carrier-grade, open-standards based 67xi SIP portfolio, offers exceptional features and flexibility in an enterprise grade IP telephone. With a sleek, elegant design and a compact footprint, this multi-line SIP telephone delivers the advanced features and performance traditionally found only in higher priced products. Featuring a 3 line LCD display, the 6730i supports up to 6 lines with call appearances, offers advanced XML capability to access custom applications and is fully interoperable with leading IP-PBX platforms.

Supported by a host of Aastra configuration options, XML development tools and continuous product enhancements via software releases, the value offered by the 6730i makes it is ideally suited for light telephone use for the small and home-based business applications.

[[Aastra_6730i_VoIP_Phone|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Atcom AG188N==

[[File:Atcom-ag188n.jpg|300px|thumb|left|Atcom AG188N]]

'''Product:''' Atcom AG188N

'''Company:''' Atcom

'''Overview:''' AG188N voice gateway is an Internet based one port voice gateway. AG188N ATA adapts multi voice control protocols and voice compression codec to directly convert analog voice into IP packet for internet transport, thus effectively using the existing bandwidth to provide PSTN quality voice service.

AG188N ATA one FXS voice over IP gateway supports SIP, and IAX2 protocol, offering one 10/100Mbps Ethernet interface, one RJ11 telephone interface, one built-in router and lifeline port. AG-188N ATA is small and can be widely used in SOHO, small office and enterprise branches.

[[Atcom AG188N|See Configuration Details]]
<div style="width:100%;overflow:hidden;clear:both"></div>

==Cisco Linksys PAP2==

[[File:Pap2.jpg|300px|thumb|left|Cisco Linksys PAP2]]

'''Product:''' Linksys PAP2

'''Company:''' Cisco

'''Overview:''' The Linksys PAP2 is an analog telephony adapter (ATA), which allows for the connection of up to two analog telephones to a softswitch or VoIP carrier using Session Initiation Protocol (SIP). The physical connections on the device are two RJ11 jacks for the telephones and one RJ45 jack for ethernet. The ethernet connection is 10 Mbit/s, half-duplex. The device is configured through a web interface.
Each analog telephone is presented as a distinct SIP user.

[[Cisco Linksys PAP2|See Configuration Details]]
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==Cisco Linksys PAP2T==

[[File:Pap2t.jpg|300px|thumb|left|Cisco Linksys PAP2T]]

'''Product:''' Linksys PAP2T

'''Company:''' Cisco

'''Overview:''' The Cisco PAP2T Internet Phone Adapter with 2 VoIP Ports enables high-quality feature-rich VoIP service through your broadband Internet connection. Just plug it into your home router or gateway and use the two standard telephone ports to connect analog phones or fax machines. Each phone port operates independently, with separate phone service and phone numbers, like having two telephone lines. You'll get clear reception and reliable fax connections, even while using the Internet.

With Internet telephony, along with low domestic and international phone rates, impressive arrays of special telephone features are available. Choose your preferred local dialing number, regardless of where you live. Or add a virtual telephone number to have forwarded to your Internet phone. You can even add a toll-free number. The Cisco PAP2T Internet Phone Adapter is compatible with these and all of the other special telephone features that are available from your Internet telephone service provider, such as caller ID, call waiting, voicemail, call forwarding, distinctive ring, and much more.

[[Cisco Linksys PAP2T|See Configuration Details]]<br>
[[PAP2T With Static IP|See Configuration Details with Static IP]]
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==Cisco SPA112==

[[File:SPA112.jpg|300px|thumb|left|Cisco SPA112]]

'''Product:''' SPA112

'''Company:''' Cisco

'''Overview:''' The Cisco SPA112 2 Port Adapter enables high-quality VoIP service with a comprehensive feature set through a broadband Internet connection. Easy to install and use, it works over an IP network to connect analog phones and fax machines to a VoIP service provider and provides support for additional LAN connections.

The Cisco SPA112 includes two standard telephone ports to connect existing analog phones or fax machines to a VoIP service provider. Each phone line can be configured independently. With the Cisco SPA112, users can protect and extend their investment in their existing analog telephones, conference speakerphones, and fax machines as well as control their migration to IP voice with an extremely affordable, reliable solution.

[[Cisco SPA112|See Configuration Details]]

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==Cisco Linksys SPA942 NA==

[[File:SPA942-0.jpg|300px|thumb|left|Cisco Linksys SPA942 NA]]

'''Product:''' Linksys SPA942 NA

'''Company:''' Cisco

'''Overview:''' Four active lines and dual switched Ethernet ports offer incredible flexibility to the home or small-business Internet telephony user. Part of Cisco Small Business IP Phone Series, the SPA942 IP Phone can be configured to carry a unique phone number or extension for each of its lines, or can be configured to use a shared number over multiple phones.

Based on the SIP standard, the SPA942 is able to easily integrate new features and services offered by providers. A high-resolution display makes for easy menu- and web-based configuration.

[[Cisco_Linksys_SPA942_NA|See Configuration Details]]
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==Cisco IP Phone 7940/7960==

[[File:Cisco_7960g.png|300px|thumb|left|Cisco IP Phone 7940/7960]]

'''Product:''' Cisco IP Phone 7940/7960

'''Company:''' Cisco

'''Overview:''' The Cisco IP Phone 7940/7960 is a full-feature telephone that provides voice communication over an IP network. This phone functions like a traditional analog phone, allowing you to place and receive telephone calls. This phone also supports features, such as network call forwarding, redialing, speed dialing, transferring calls, placing conference calls, and accessing voice mail.

[[Cisco_IP_Phone_7940/7960|See Configuration Details]]
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==Cisco SPA2100 Phone Adapter==

[[File:Sipura2100.jpg|300px|thumb|left|Cisco SPA2100 Phone Adapter]]

'''Product:''' Cisco SPA2100 Phone Adapter

'''Company:''' Cisco

'''Overview:''' Inexpensive, easy to install, and simple to use, the Cisco® SPA2100 Phone Adapter with Router (Figure 1) connects a standard telephone or fax machine to an IP-based data network. Voice over IP (VoIP) service providers can offer residential and business users traditional and enhanced communication services via the customer's broadband connection to the Internet.

The Cisco SPA2100 features two basic telephone ports to connect existing analog phones or fax machines to a private branch exchange (PBX) system. It also includes two RJ-45 Ethernet interfaces to connect to a home or office LAN, as well as an Ethernet connection to connect a broadband modem or router (WAN). Each phone line can be configured independently via software controlled by the service provider or the end user. With the SPA2100, users are able to protect and extend their investment in their existing analog telephones, conference speakerphones, and fax machines, as well as control their migration to IP with an extremely affordable, reliable solution.

[[Cisco SPA2100 Phone Adapter|See Configuration Details]]
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==Cisco SPA2102 Phone Adapter with Router==

[[File:Spa2102b.jpg|300px|thumb|left|Cisco SPA2102 Phone Adapter with Router]]

'''Product:''' Cisco SPA2102 Phone Adapter with Router

'''Company:''' Cisco

'''Overview:''' Inexpensive, easy to install, and simple to use, the Cisco SPA2102 Phone Adapter with Router connects a standard telephone or fax machine to an IP-based data network. Voice over IP (VoIP) service providers can offer residential and business users traditional and enhanced communication services via the customer's broadband connection to the Internet.

The Cisco SPA2102 features two basic telephone ports to connect existing analog phones or fax machines to a private branch exchange (PBX) system. It also includes two 100BASE-T RJ-45 Ethernet interfaces to connect to a home or office LAN, as well as an Ethernet connection to connect a broadband modem or router (WAN). Each phone line can be configured independently via software controlled by the service provider or the end user. With the SPA2102, users are able to protect and extend their investment in their existing analog telephones, conference speakerphones, and fax machines, as well as control their migration to IP with an extremely affordable, reliable solution.

[[Cisco_SPA2102_Phone_Adapter_with_Router|See Configuration Details]]
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==Cisco SPA504G Phone==

[[File:Cisco_spa504g.jpg|300px|thumb|left|Cisco SPA504G Phone]]

'''Product:''' Cisco SPA504G Phone

'''Company:''' Cisco

'''Overview:'''
The Cisco SPA504G uses standard encryption protocols to perform highly secure remote provisioning and
unobtrusive in-service software upgrades. Remote provisioning tools include detailed performance measurement
and troubleshooting features, enabling network providers to deliver high-quality support to their subscribers. Remote
provisioning also saves service providers the time and expense of managing, preloading, and reconfiguring
customer premises equipment.

[[Cisco_SPA504G_Phone|See Configuration Details]]
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==Grandstream HandyTone 286==

[[File:Grandstream286.gif|300px|thumb|left|Grandstream HandyTone 286]]

'''Product:''' Grandstream HandyTone 286

'''Company:''' Grandstream

'''Overview:''' Grandstream's award-wining HandyTone-286 is innovative Analog Telephone Adaptor that offers a rich
set of functionality and superb sound quality at ultra-affordable price. They are fully compatible with SIP
industry standard and can interoperate with many other SIP compliant devices and software on the
market

[[Grandstream_HandyTone_286|See Configuration Details]]
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==Grandstream HandyTone 486==

[[File:HT486.jpg|300px|thumb|left|Grandstream HandyTone 486]]

'''Product:''' Grandstream HandyTone 486

'''Company:''' Grandstream

'''Overview:''' Grandstream's award-wining HandyTone-486 is an all-in-one VoIP integrated access device that features superb audio quality, rich functionalities, high level of integration, compactness and ultraaffordability. The HandyTone-486 is fully compatible with SIP industry standard and can interoperate with many other SIP compliant devices and software on the market.

Grandstream HandyTone-486 has been awarded the Best of Show product in 2004 Internet Telephony Conference and Expo.

[[Grandstream_HandyTone_486|See Configuration Details]]
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==Grandstream HandyTone 502==

[[File:Ht502.jpg|300px|thumb|left|Grandstream HandyTone 502 - HT502]]

'''Product:''' Grandstream HandyTone 502 - HT502

'''Company:''' Grandstream

'''Overview:''' The HT502 is a powerful VoIP router. The product's inclusion of an integrated high performance NAT router and 10Mbps Ethernet WAN and LAN ports enables a shared broadband connection between multiple ethernet devices.

In addition to being SIP 2.0 standard compliant, the product supports Universal Plug-in-Play (UPnP), up to 2 SIP account profiles, and advanced telephony features.

Enhanced security

Automated provisioning using symmetric and asymmetric voice

Support for a broad range of popular voice codec

[[Grandstream_HandyTone_502_-_HT502|See Configuration Details]]
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==Mediatrix 4100 Series==

[[File:Mediatrix4102s.jpg|300px|thumb|left|Mediatrix 4102]]

'''Product:''' : Mediatrix 4102S and 4100 Series running DGW 2.0 firmware

'''Company:''' Media5 Corporation

'''Overview:''' The Mediatrix® 4100 Series device is a security ready VoIP gateway, connecting up to two analog phones and/or faxes, as well as a PC or a home router to a broadband modem. The Mediatrix 4102 offers security features such as SIP over TLS, SRTP, certificates management, and HTTPS designed to bring enhanced security for the network management, SIP signaling and media transmission aspects.

[[Media5 Mediatrix 4100|See Configuration Details]]
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==Netgear WGR615V==

[[File:NetgearWGR615V.jpg|300px|thumb|left|Netgear WGR615V]]

'''Product:''' WGR615V

'''Company:''' Netgear

'''Overview:''' The WGR615V was never sold to directly to the public when it was new but since the model is discontinued, lots of them are being sold.
It is a wireless (802.11 G) router that happens to have a built-in ATA.
It is possible to use just the ATA part.

[[Netgear WGR615V|See Configuration Details]]
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==OBi110==

[[File:OBi110-ATA.jpg|300px|thumb|left|OBi110]]

'''Product:''' OBi110

'''Company:''' OBIHAI Technology Inc

'''Overview:''' With OBi devices you are in control of your communications life. From the OBi's on-board connections as well as via the Internet, you have the power to bridge mobile, fixed line and Internet telephone services. The OBi100 & OBi110 are stand-alone, dedicated devices, built with a high-performance "system on a chip" platform to ensure high-quality voice conversations. Every OBi device has high availability and reliability. It is 'always-on' to make or receive a call. With an OBi device, a computer is not required and a computer does not need to be on to talk on the phone or use the OBi's powerful service bridging features. To get started, all you need is a phone, power and a connection to the Internet.

The OBi110 has one phone port and one line port. Use the OBi110 when you need an analog line connection to a traditional telephone network or service. Think of the OBi110 LINE port as a gateway to a traditional phone service. If you do not have a traditional phone service at home, then the OBi100 is probably the right product to get.

[[OBi110|See Configuration Details]]
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==Panasonic KX-TGP 550==

[[File:Pana550b.jpg|300px|thumb|left|Panasonic KX-TGP 550]]

'''Product:''' Panasonic KX-TGP 550

'''Company:''' Panasonic

'''Overview:''' Panasonic KX-TGP 550 SIP Cordless Phone System allows you to have up to eight (8) phonenumbers.You can set up in several ways: for example, you can set thephone number for each handset. Or you can group the handsets bygroup setting and restrict the incoming calls receivals to the specifichandsets. Handsets if you need them.

*Support for 3 simultaneous network conversations
*CODEC: G.711a-law / G.711μ-law / G.722(wideband) / G.729a / G.726(32K)
*DECT radio technology
*2.1" Large LCD with white backlight on cordless handset
*Up to 6 DECT cordless handsets*1
*Support for up to 8 SIP registrations (e.g. up to 8 DID lines or extensions)
*Hands-free speaker phone on cordless handset
*Wall mountable base unit

[[Panasonic KX-TGP 550|See Configuration Details]]
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==Pirelli DP-L10==

[[File:Pirelli1.jpg|300px|thumb|left|Pirelli DP-L10]]

'''Product:''' Pirelli DP-L10

'''Company:''' Pirelli

'''Overview:''' Pirelli's DualPhone DP-L10 phone combines GSM tri-band capabilities with SIP-based Wireless VoIP via an integrated WLAN 802.11b/g interface, combining cell phone technology with the added advantage of transporting phone calls over the Internet.

The Dual Phone enables mobile phone features such as Internet browsing, e-mail, SMS, and MMS, which can also be used through a fixed-broadband connection, providing the same mobile services with the added benefits of greater bandwidth and cost savings.

[[Pirelli DP-L10|See Configuration Details]]
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==Polycom SoundStation IP 4000 Conference Phone==

[[File:Polycom-4000-0.gif|300px|thumb|left|Polycom SoundStation IP 4000 Conference Phone]]

'''Product:''' Polycom SoundStation IP 4000 Conference Phone

'''Company:''' Polycom

'''Overview:''' the SoundStation IP 4000 SIP voice conferencing unit is the answer for organizations that are ready for the
benefits and versatility of a SIP enabled business. Designed for offices or small to medium sized conference rooms, the SoundStation IP 4000 SIP provides remarkable room coverage. You can speak naturally from up to 10-feet away from a microphone and still be heard clearly on the far end of the call. Need coverage for a larger room? The optional extension microphones offer an increased pickup for larger rooms. Plus, with gated microphone technology, echo and background noise is almost entirely eliminated.

The SoundStation IP 4000 SIP also provides familiar features that are easy to use. The menu driven user interface, viewable on a high-resolution backlit LCD, offers convenient access to business telephony functions you use most including transfer, hold, redial, and conference.

[[Polycom_SoundStation_IP_4000_Conference_Phone|See Configuration Details]]
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==Polycom SoundPoint IP 501, 550, 650, etc.==

[[File:Polycom501.jpg|258px|thumb|left|Polycom SoundPoint IP 501]]

'''Product:''' Polycom SoundPoint IP 501, 550, 650, etc.

'''Company:''' Polycom

'''Overview:''' The SoundPoint IP series are multi-line (3-6 lines depending on model) Voice over IP telephones that seamlessly integrate with IP PBX and softswitch vendors’ IP solutions.

As protocols develop and standards evolve, it’s easy to update the phone in the field via a software download, thus enabling new features and functionality for the phone and protecting your investment.

Whether you deploy MGCP or SIP standards, Polycom offers a solution that fits all your business communication needs.

[[Polycom_SoundPoint_IP_501|See Configuration Details]]
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==Siemens Gigaset C450-Ip==

[[File:Siemens_gigaset_c450_IP.jpg|300px|thumb|left|Siemens Gigaset C450-Ip]]

'''Product:''' Siemens Gigaset C450-Ip

'''Company:''' Siemens

'''Overview:''' The Gigaset C450 is an entry phone in the Siemens Gigaset product line, the addition of IP connectivity via a standard Ethernet 10/100 BaseT interface and SIP protocol within a reasonable price bracket, opens large applications where wireless operation and SIP are required.
The Ethernet connectivity allows the unit to work without a PC.

The dual mode (SIP/PSTN) enables incoming call on legacy interface with existing directory number and safe emergency outgoing calls.

The presentation of the unit with an independent base and a dedicated phone charger/support will simplify cabling.

[[Siemens Gigaset C450-Ip|See Configuration Details]]
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==SNOM 320 VoIP Phone==

[[File:Snom320.png|300px|frame|left|Snom 320 VoIP Phone]]

'''Product:''' Snom 320 VoIP Phone

'''Company:''' Snom

'''Overview:'''Ideal for the office and everyone who spends a lot of time on the phone, the snom 320 is an affordable, yet powerful SIP business phone with built-in, full-duplex speakerphone and three-party conference bridging.

[[SNOM 320|See Configuration Details]]
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==Snom m3 VoIP Phone==

[[File:Snom_m3.png|300px|thumb|left|Snom m3 VoIP Phone]]

'''Product:''' Snom m3 VoIP Phone

'''Company:''' Snom

'''Overview:''' The snom m9 is the next generation DECT handheld (CAT-iq) that empowers users with the convenience of wireless communication along with the widely accepted benefits and feature richness of Voice-over-IP telephony.

The snom m9 promises to deliver excellent speech quality through digital and wideband audio (klarVOICE).

By combining professional functions of versatile business communication with the intuitive features of the mobile-carrier world, the snom m9 is ideally suited for professional and private use alike. With features such as hands free mode, calling line identification (CLI) by displaying name, number and image of the caller as well as typical mobile-phone features such as Address Book, Calendar, Calculator and Alarm function, the snom m9 provides the perfect blend of mobility and accessibility.

[[Snom_m3_VoIP_Phone|See Configuration Details]]
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==Telco AC-211==

[[File:Ac211n.jpg|300px|thumb|left|Telco AC-211]]

'''Product:''' Telco AC-211

'''Company:''' Telco

'''Overview:''' Telco AC-211 is a SIP ATA supporting two lines and numerous configuration options. This ATA is most commonly found as the AC-211-SR & AC-211N from the now defunct SunRocket service. This device works well with voip.ms once configured.

[[Telco_AC-211|See Configuration Details]]

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==Yealink SIP-T28P (VSRF)==

[[File:Yealink-sip-t28p.jpg|300px|thumb|left|Yealink SIP-T28P]]

'''Product:''' Yealink SIP-T28P

'''Company:''' Yealink

'''Overview:''' Yealink SIP-T28P represents the next generation VoIP phone which designed for the business user who needs rich telephony features, friendly UI and super voice quality. It is equipped with the TI TITAN chipset, offers high definition voice quality through TI voice engine, HD handset, HD speaker and HD codec (G.722).

By the large, high-resolution graphical display, and together with all the 48 keys, SIP-T28P offers an excellent user experience to configure, make calls, express XML browser, etc. Moreover, to avoid problems with unwanted violations of your audio data, Yealink SIP-T28P supports the security standards TLS, SRTP, HTTPS, 802.1x, Open VPN and AES encryption which are necessary to protect against electronic eavesdropping and data theft.

[[Yealink_SIP-T28P|See Configuration Details]]
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[[Category:Devices]]

==Zycoo ZP502==

[[File:ZP502.jpg|left|Zycoo ZP502]]

'''Product:''' Zycoo ZP502

'''Company:''' Zycoo ZP502

'''Overview:''' ZP502 VoIP Phone is ZYCOO business IP phone terminal which adopts SIP and IAX2 protocols.

With Broadcom solution,compatible with various Platforms such as Asterisk, FreePBX, Broadsoft, Cisco call manager etc.

[[Zycoo ZP502|See Configuration Details]]
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Media5-fone

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==Configuration Setup==

[[File:Media5Android.jpg|250px]]

'''Download the Media5-fone from the Android Market or the Apple App Store'''

*'''Start the application'''
-'''Click on More'''
-'''Settings'''
-'''Configure SIP Accounts'''

[[File:Media5Android2.jpg|250px]]

*'''Click on the + sign on the top right corner'''
-'''Click on Internet Telephony Service Providers, Preconfigured List'''

[[File:Media5Android3.jpg|250px]]

*'''Select one of the VoIP.ms servers.'''

[[File:Media5Android4.jpg|250px]]

*'''Configure your VoIP.ms username:''' 100000 (Replace with your VoIP.ms username)
*'''Configure your VoIP.ms Password:'''(Your account password)

[[File:Media5Android5.jpg|250px]]

*Click the '''DONE''' button.
*You should be ready to make calls