VoIP.ms Wiki - User contributions [en]

Registration issue

2017-01-05T16:31:28Z

Devnull: /* Can`t Connect or Register to Login Server */ updated solutions to suggest port change

Sometimes it is possible to have issues when trying to register your [http://wiki.voip.ms/article/Devices Device] / [http://wiki.voip.ms/article/Softphones Softphone] with VoIP.ms, we will review some recommendations and important things to check, to start trouble shooting with this situation.

How can we notice there is an issue with registration? Some indicators that can alert us to it:

*When I try to make a call, there is '''no dial tone''' on the line.
*I used to have a working DID number and '''I no longer receive calls''', or calls are going directly to my voicemail or to a failover.
*At the customer portal I can not see my '''SIP Registration''' (Note: This does not apply for IP Authentication).

It is possible that some specific situations will need to be addressed with [http://voip.ms/ VoIP.ms] staff, but with this article you can start looking for potential causes and solutions for your issue.

__TOC__

== Reboot Your Devices ==

Most common issues related with registration failing, can be resolved just by rebooting the [http://wiki.voip.ms/article/Devices Device]/[http://wiki.voip.ms/article/Softphones Softphone], the router and the modem. This suggestion can usually help in the case you had a power outage or if your Internet connection stopped working for a while.

Note: Considering the use of failover options can help to prevent losing calls during the trouble shooting, you can use a call forward or a
[[voicemail]] along with your number.

== ATA/[http://wiki.voip.ms/article/Softphones Softphone] ==

Some common suggestions that can be followed if the issue is related with an ATA device or a [http://wiki.voip.ms/article/Softphones softphone]:

*Verify if your device has any field to set the '''"Register expires"''', if you find something like this, you usually will see a default of 3600 (seconds), lower this value to 120 (2 mins). This tells our side that you are no longer registered and then failovers will kick in immediately.

* '''Register Expires''' is the parameter that controls how often your client contacts the SIP server to remind it that the client is alive and confirming its current location (public IP address and listening SIP port). The SIP server is supposed to set this timer as part of the reply to each Register command. If the automated setting doesn't work you need to set this parameter manually according to the provider's instructions.

*Use the '''IP address''' from the server instead of the domain name, example: Use 67.215.241.250 instead of losangeles.voip.ms. ''This item is recommended as diagnostic only, if this works for you, then it is probably a DNS issue affecting in your network, test using it for a couple of weeks and then change again to the domain name, it is not recommended to have the registration with the Ip all the time, VoIP.ms will redirect the domain to another IP in the case of a server issue, but this will not work for you if you are not using the domain.''

* If your device has any options similar to '''"Nat Keep Alive"''' or '''"Nat Mapping"''' please enable them. This will make sure the connection does not go idle.

*Check if you have the '''Latest Firmware''' updated on your device, some minor bugs get resolved with the new firmwares, so it is important to have them up to date.

== Check Ports ==

By default is not necessary to open ports in the router and this is only required in the case of some specific firewall rules are blocking traffic on those ports, however if issues are present, we can forward '''ports UDP 5060 and also 10000 to 20000 UDP''', in the router. Another suggestion is when using different devices under the same network, you can designate an internal port for each one (for instance using port 5061 in one device, 5062 in next one, and so on). Communication between the VoIP.ms and the router will still go through '''port 5060''', even though different internal ports are used for each device.

Do you suspect your ISP could be blocking these ports ? It is recommended to contact them just to be certain.

If you have a device capable of supporting IAX2 protocol, you can use this as an alternative, just make sure port UDP 4569 is not blocked in your network.

== Can`t Connect or Register to Login Server ==

Check if your device shows any information related with the error when it can not connect, for example on some ATA [[devices]], you can see the "Registration Status", if it shows something like '''"Cant connect to login server"''' or '''Failed''', you can follow these suggestions:

*This error is usually caused by the '''router''' that somehow is blocking the registration attempts, try rebooting the router then the ATA device, if that does not work, re-enter the password and UserID on the device settings (Please check this Page for [https://www.voip.ms/m/samples/voxalot.php Main Login Credentials]), or try using the IP address of the server instead of the hostname. If you do not have the IP address send a ping to the server or contact the [http://voip.ms/ VoIP.ms] staff.

* If this is a new installation and you are just trying to get the service working, then if at all possible please remove your router from the loop. Connect your ATA Device / Computer directly to the Internet without the router in between. If you can then register and receive / make calls you will know there is a router configuration problem preventing proper connectivity.

* You can attempt to use a different SIP port, [[FAQ#Do_you_offer_alternative_ports_besides_5060.3F|such as <tt>5080</tt> and <tt> 42872</tt>]]

== Tomato Firmware (PAP2TNA). ==

This part is for PAP2T devices, when using the '''Tomato Firmware''':

*Some of the default settings with Linksys devices conflict with the default Tomato timeout settings. These are the UDP Timeout settings and if they are left at their defaults, VoIP devices will sometimes fail to register after an IP change. These may be found on the '''Conntrack/Netfilter page''' in the Tomato config. Set '''"unreplied timeout"''' to 10 seconds.

== Asterisk /[http://wiki.voip.ms/article/PBXs PBX] system. ==

Some common suggestions that can be followed if the issue is related with an Asterisk system or a PBX:

*Add to your trunk '''nat=yes''' and '''qualify=yes''', these 2 values can help with your registration issues.

*Use the '''IP address''' from the server instead of the domain name, example: Use 67.215.241.250 instead of losangeles.voip.ms. ''This item is recommended as diagnostic only, if this works for you, then it is probably a DNS issue affecting in your network, test using it for a couple of weeks and then change again to the domain name, it is not recommended to have the registration with the IP all the time, VoIP providers tend to redirect the domain to another IP in the case of a server issue, but this will not work if you are not using the domain.'

=== Nat=yes ===

The '''NAT option''' determines the type of setting for users trying to connect to an asterisk server. With [http://voip.ms/ VoIP.ms] it is recommended to have the NAT option set on Yes, which is the option that will work best.

Portal settings influencing NAT with Asterisk:

*'''yes = Always ignore info and assume NAT'''
*no = Use NAT mode only according to RFC3581 (;rport)
*never = Never attempt NAT mode or RFC3581 support
*route = Assume NAT, don't send rport (work around more UNIDEN bugs)

=== Qualify=yes ===

This feature may also be used to keep a '''UDP session open''' to a device that is located behind a network address translator (NAT). By sending the OPTIONS request, the UDP port binding in the NAT (on the outside address of the NAT/firewall device) is maintained by sending traffic through it. If the binding was to expire, there would be no way for Asterisk to initiate a call to the SIP device. This can be used in conjunction with the nat=yes setting.

Note that it is the qualify=xxx(in miliseconds) or qualify=yes or qualify=no in sip.conf file that determines the registration with the
server. If you have "yes" for qualify it will check the registration with the server every 2 seconds.

[[category:guides]]

SMS

2016-10-11T16:09:36Z

Devnull: /* Send and Receive Messages */ added in section re: iOS/android apps

[[File:Smslogo4.jpg|200px|left]]

Short Message Service (SMS). This feature will allow you to send and receive messages with your DID Number (US and Canada DID Numbers Only). Currently it is in beta and will remain free until further notice.

Please note that this feature is for regular customer usage. '''No automation, telemarketing, bulk sending or receiving will be allowed.'''

__TOC__

== Important information to know about the SMS Service ==

* '''The SMS Service is in BETA version, that means it is not fully deployed. It is important to us that you report any issues with this service by sending an email to support@voip.ms so that the developers can get involved if necessary.'''

* '''The SMS Service will be free until further notice.'''

* '''The SMS Service is only available for US and Canadian local DID Numbers marked with the distinctive SMS Icon.'''

* '''The SMS Service can only be used to send SMS Messages to Canadian and US 10 digit numbers at this time. We also cannot guarantee that International SMS will be properly received.'''

* '''VoIP.ms reserves the right, at its sole discretion, to limit or disable the SMS service from any account that may present marketing patterns / automation patterns / bulk sending or receiving while the service is free and in beta.'''

* '''At this time we cannot guarantee that Short Code SMS Messages, which are usually 6 digits or less (e.g.: Skype, Bank Codes, TV Commercials etc...), will work.'''

* '''We cannot guarantee that accents or special characters including non Latin letters will be properly delivered.'''

* '''For Ported In Numbers: The SMS Functionality will be available up to 48 hours after the porting process is marked as Completed.'''

* '''IMPORTANT: At this time, it is not possible to use a softphone to send or receive messages with our SMS Feature. This has to be done through the SMS Message Center in the Customer Portal at DID Numbers >> SMS Message Center in the Customer Portal or through https://sms.voip.ms

If you have further questions don't hesitate to contact the Support Staff on the Live Chat or Ticket System.

== Identifying a SMS DID Number ==

The first thing you need to know is that this feature is only available for '''local US and Canadian numbers''' at the moment. You will note that some numbers will have a little icon of a phone device, this indicates that the number supports SMS. Please note that not all locations support SMS at this time.

You can start ordering a DID Number from your portal >> DID Numbers >> Order DID >> Local Numbers >> (US or Canada) Numbers and you will be able to purchase your desired number, just make sure it supports the SMS feature.

[[File:SMS 1.png]]

== Configuring the SMS service ==

Once purchased the SMS Service needs to be activated on the DID settings, from your portal >> DID Numbers >> Manage DID >> Edit Selection >> SMS.

On that section you can activate and configure some forward options for the SMS service.

[[File:SMS2.png]]

The first action to take is to mark the '''Short Message Service (SMS) service''', by enabling this field the SMS Service will be active and free until further notice.

* '''VoIP.ms SMS Portal''': This is enabled by default and it can't be modified, that means that you will be able to create and send messages from the SMS Message Center.

*'''SMS Email Address''': By activating this option all the SMS Messages will be sent to the email address you configure in this field. The advantage of this option is that you will receive your SMS Messages directly to your email and you will be able to Reply to these messages from your email too. You just need to click on '''reply''' to the email.

To reply the SMS Message via your email it is important to click on Reply and '''DON'T change or modify''' the destination mail "TO"
(sms@voip.ms)and the subject E.G [#USXXX] Message sent to 5555555555.

*'''SMS Forward''': You can have your SMS Messages forwarded to your cellphone or any other number that supports SMS, with this option you will receive the SMS Messages from our system to the configured phone number.

The CallerID sent to the Call Forwarded Cell Phone will be your DID Number that received the SMS Message.

*'''SMS URL Callback''': By enabling this option you will be able to send the messages to another destination e.g. your own server. If selected SMS Messages received by your DID will Send a GET request to the URL Callback provided. Available variables for your URL:

'''{ID}''' The ID of the SMS message.
'''{TIMESTAMP}''' The date and time the message was received.
'''{FROM}''' The phone number that sent you the message
'''{TO}''' The DID Number that received the message
'''{MESSAGE}''' The content of the message
'''Example''': http: //mysite.com/sms.php?to={TO}&from={FROM}&message={MESSAGE}&id={ID}&date={TIMESTAMP}

*'''URL Callback Retry''': When selected, we will be expecting an "ok" output (without quotes) from your URL callback page as an indicator that you have received the message correctly. If we don't received the "ok" (without quotes) from your callback page, we will keep sending you the same message every 30 minutes.

Once you configure the desired option, click on Apply Changes.

== Send and Receive Messages ==

===SMS Message Center===

To start using the service you will need to use the SMS Message Center, from your portal >> DID Numbers >> [https://www.voip.ms/m/sms.php SMS Message Center].

For Mobile Devices please use [https://sms.voip.ms/ sms.voip.ms] as a complete all in one solution.

From the SMS Message Center you can check your SMS History, send and receive SMS messages.

[[File:SMS.png]]

*'''Search Range''': Select a Date Range of your SMS History. You can select a time period by marking the "Show Details" field and the text messages will be displayed.

*'''Search Filter''': You can filter your Search by DID, Contact and Type (Sent, Received and Both).

*'''Send New SMS Messages''': Another window will open by clicking here in order to create a new SMS Message.

*'''Delete Selected Messages''': This action will delete any messages that you have selected, by clicking on the box next to them. '''This action cannot be undone.'''

*'''Delete All Messages''': Delete ALL your SMS Messages, both sent and received. '''This action cannot be undone.'''

*'''Checking a Message''': To check a received SMS Message just click it directly in order to display the information.

====Create a New SMS Message====

[[File:SMS41.png]]

You can create a new '''SMS Message''' we will explain you all the parameters you can find on this section:

* '''DID''': If you have more than one DID Number you can select any of the numbers available to send your message.
* '''Contact''': The destination phone number or if you have a [[Phone book|Phone book entry]] start typing the name and the system will display the information if available.
* '''Message''': You can create your message on that field, please note that you have up to 160 characters.

Then click on 'Send Message' to send the SMS.

=== VoIP.ms SMS via Native Android/iOS Application ===

Community effort have produced a pair of native apps that expose SMS to phones running Android and iOS. The apps themselves can be retrieved here:

* [https://play.google.com/store/apps/details?id=net.kourlas.voipms_sms&hl=en Google Play Store]
* [https://itunes.apple.com/ca/app/voip.ms-sms/id1037874743?mt=8 App Store]

These apps work by interfacing with your VoIP.ms account via their API interface. By default, this interface is disabled, and must be enabled (Main Menu -> Soap and REST/JSON API):

[[File:Api-config.png]]

* '''API Password''': Create your own secure password in this field
* '''Enable IP Addresses''': If you'd like to lock down access to the API to a specific set of IP addresses, define them here. The special value <code>0.0.0.0</code> can be used to allow the API to respond to all requests, irrespective of their origin IP address.

Enable the API interface, and then configure the VoIP.ms SMS app to use connect to your account.

File:Api-config.png

2016-10-11T16:06:36Z

Devnull: VoIP.ms API configuration page

VoIP.ms API configuration page

Cisco SPA112

2016-10-07T19:21:51Z

Devnull: Added instructions on finding device IP via router

[[File:SPA112.jpg|300px|thumb|left|Cisco SPA112]]
<div style="width:100%;overflow:hidden;clear:both"></div>

: '''There have been some reports of issues with this device, from both customers of VoIP.ms and other providers.'''
: '''Make sure to install the latest firmware from [https://software.cisco.com/download/release.html?mdfid=283998771&softwareid=282463187&release=1.4.0&relind=AVAILABLE&rellifecycle=&reltype=latest Cisco Software].'''
: '''Version 1.1 or later should be used for proper Caller ID support. '''
: '''Some People have reported issues using Firefox to Configure this device please try Chrome or IE. '''

== Configuration Details ==

=== Getting the IP address of your device ===

There are two ways to retrieve the IP address of your Cisco SPA112, via analog phone menu, and via your internet router.

{| class="wikitable"
! style="text-align: center;" | Analog phone interface
! style="text-align: center;" | Internet Router
|-
|
# Attach the Cisco SPA112 to your network and attach an analog phone to one of the phone ports, then do the following:
## Dial **** from the phone, even though there is no dial tone.
## When you hear "System Configuration Menu," dial 1 1 0 # slowly. The current IP address will be read back. (e.g. 192.168.X.X)

'''If you hear 0.0.0.0, check your network connection and DHCP server. If necessary, a static IP address'''
'''can be assigned by using option 111# at the IVR, then entering the IP address with your phone's keypad'''
'''(for example, 10*1*27*2 for 10.1.27.2). The network mask can be set with option 121# and the default'''
'''gateway can be sent with option 131#'''
Learn more about the IVR menu options from the https://supportforums.cisco.com/docs/DOC-9900 document.

\Be sure to allow at least a minute or two for the box to initialise; even a correctly configured and installed SPA112/122 will give no power to the 'phone or no dialtone until initialisation is complete.

Note that the SPA122 is basically a SPA112 with a second network port, intended for installation between a local network hub (LAN) and an upstream Internet (WAN) connection. The SPA122 may be configured as either a "NAT" or "bridge". Depending on configuration, this leaves the SPA122 with two addresses; a local area network address (such as 192.168.15.1) and an outside Internet address. Dialling ****110# will give one address, ****210# will give the other.
|
# Attach the Cisco SPA112 to your network
# Access your router's remote administration interface via your web browser, typical addresses may be <code>192.168.0.1</code> or <code>192.168.1.1</code>. Refer to your router instructions for more information.
# Enter your username/password if challenged. If you have not set one, then it is likely the unchanged default password.
# In the router's menu, there should be a page showing a list of connected clients, with their internal IP address. Find the entry corresponding to the Cisco SPA112. It should identify itself in the list as "SPA112"
# Navigate to this IP address via your web browser
|}

=== Accessing to the device's settings page ===

Open your web browser and go to the IP address you obtained in step 1 (for example, http://192.168.2.1).
The default username is admin, and the default password is also admin.

For the SPA122, if one address does not return the web interface (or has some functions greyed/disabled), try the other.

=== Configuring the Quick Setup screen ===

Go to Quick Setup and configure Line 1 as follows:

'''Proxy:''' atlanta.voip.ms (You can choose any of our multiple VoIP.ms [http://wiki.voip.ms/article/Choosing_Server servers])

'''Display Name:''' Your name

'''User ID:''' 100000 (Replace with your 6 digit Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)

'''Password:''' Your VoIP.MS SIP Password

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

(note: Replace 555 in the dial plan with your area code, See [[Dial Plan for Linksys ATAs]] for details.)

Click Submit to save settings.

[[File:quick_setup_test.png|800px|thumb|left|Quick Setup Page - click to enlarge]]
 

=== Configuring the Voice Line ===
==== Nat Settings ====

Click on Voice, then Line 1

Set '''NAT Mapping Enable''' to Yes, then set '''NAT Keep Alive Enable''' to Yes. If your environment does not use NAT, you can leave these settings disabled. These features can usually be disabled on the SPA122 if it is connected directly to your modem since its traffic will not be subject to NAT in this configuration.

[[File:VL_1_nat_settings.png|800px|thumb|left|NAT Settings - click to enlarge]]
 

==== Proxy and Registration ====

Under '''Proxy and Registration''' set the server you will use as registration server and the proper values for the register expires and proxy Fallback Intvl:

'''Proxy''': atlanta.voip.ms (one of VoIP.ms multiple [http://wiki.voip.ms/article/Choosing_Server servers], you can choose the one closest to your location.)
'''Register Expires''' to 300
'''Proxy Fallback Intvl''' to 300

Also confirm the following settings: 
'''Register:''' YES 
'''Use DNS SRV:''' NO 
'''DNS SRV Auto Prefix:''' NO

[[File:VL_2_proxyAndRegistration.png|800px|thumb|left|Proxy and Registration - click to enlarge]]
 

Click Submit to submit these changes

==== Subscriber Information ====

In this section please confirm that you have the proper account information:

'''Display Name''': Your name (that will be shown as callerID name)
'''User ID''': 100000 (Replace with your 6 digit Main SIP account UserID or Sub Account user name, i.e. 123456 or 123456_sub)
'''Password''': Your VoIP.ms SIP Password

[[File:VL_3_subscriberInformation.png|800px|thumb|left|Subscriber Information - click to enlarge]]
 

==== Audio Configuration ====

You can verify or change the audio codec that will be used with the calls. Please verify that you have the same codec selected in your SIP account's settings.

Preferred codec: g711u (or G729)

[[File:VL_4_audioConfig.png|800px|thumb|left|Audio configuration - click to enlarge]]
 

==== Dial Plan ====

We recommend to use this dial plan.

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

[[File:VL_5_dialPlan.png|800px|thumb|left|Dial Plan - click to enlarge]]
 

You can create your own dial plan if you need it, referring to this entry [[Dial Plan for Linksys ATAs]]

=== Optional settings ===

==== Outbound audio "breaking up". ====

Cisco's defaults (SIP T1 = 0.5 sec, RTP packet size 0.030 on most Sipura adapters) respectively may cause unnecessary retransmission of commands over connections with high latency and create issues with outbound audio "breaking up".

Click '''Voice''', then go to '''SIP'''.

Set SIP Timer Values (sec)

SIP T1: 1

Set RTP Parameters

RTP Packet Size: 0.02
RTP Port Min: 10000
RTP Port Max: 20000

Click Submit to save the changes

[[File:VS_sipAndRTP.png|800px|thumb|left|SIP Values - click to enlarge]]
 

==== Caller ID display showing incorrect time ====

Sometimes the hour shown in your caller ID is incorrect. Following this suggestion usually solves the issue:

Enter your device's settings and click '''Network Setup''', then go to '''Basic Setup''', then click '''Time Settings'''

Set your time zone and (optional) NTP settings. A good time server choice is 0.pool.ntp.org. Setting the proper time zone will ensure that the time which appears on your Caller ID display is correct.

Click Submit to save the changes

==Known Issues==

=== '''Phone will not ring on handset''' ===

Sometimes the Phone you are using is designed for a certain Voltage and Ring Waveform. If someone tries to call you and the phone appears to be ringing for the caller but your phone never rings please follow these steps to hopefully resolve this issue for you.

Step 1: First access the SPA web interface.

Step 2: Click on the '''Admin Login''' and then click on '''(switch to advanced view)'''

Step 3: Click on your Regional Tab on the Top Menu.

Step 4: Go Halfway Down the Page until you see the Heading '''Ring and Call Waiting Tone Spec'''

[[File:Pap2Ring.jpg|800px|thumb|left| Ring and Call Waiting - click to enlarge]]
 

Step 5: Change the Ring Waveform setting to Sinusoid or Trapezoid, the opposite of what you have set. You can also change the Ring Voltage in increments of 5 to 90 or 95.

Step 6: Save Settings and Test an Incoming Call

=== Receiving Unwanted Calls in the middle of the Night ( i.e. CallerID 100) that do not appear in your CDR: ===

These calls are not going through our Network but rather through the internet directly to your ATA Device.

Please look under the Voice>> Line 1 page in your SPA device for the following setting: Restrict Source IP and make sure it's enabled.

This way the ATA device will block any traffic not coming from our servers.

[[File:VL_1_restrictSourceIP.png|800px|thumb|left|Restrict IP - click to enlarge]]
 

=== Firmware Upgrade ===

SPA112 and SPA122 adapters were distributed with outdated (1.0.x) firmware at least as late as 2012; affected boxes will not show Caller ID on any inbound call, even though the caller names and numbers are visible in the call detail record on the VoIP.ms (or other provider's) web interface.

Updated firmware is available from the Cisco site [https://software.cisco.com/download/release.html?mdfid=283998771&softwareid=282463187&release=1.4.0&relind=AVAILABLE&rellifecycle=&reltype=latest Cisco Firmware] as a .ZIP archive which contains two files (a .BIN with the actual firmware and a .PDF with documentation). Download and unZIP this file. Go to the 'administration' tab on the web interface (on the SPA122, this needs to be done from the LAN side with SPA122's built-in networking set to NAT mode). On the left sidebar, click 'update firmware' (as most of the administration menu does not appear for Firefox users, downgrade to MS IE or another browser temporarily). Click the 'upload' button and indicate the location of the unzipped .BIN file. A box will appear with a progress indicator and a warning not to interrupt the upgrade. When the upgrade is completed, the SPA112/122 will reset and will likely take a minute or more to reinitialize, reconnect to the network and restore dial tone. SPA122 users who have installed the device in-line between the local PCs and the Internet will be disconnected from the Internet until reinitialization is complete.

Once the new firmware is deployed, call display will operate normally and the configuration web page will display in Firefox without missing options in the administration menu.

A manual for Cisco's SPA100 series adapters is online at http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/spa100-200/admin_guide_SPA100/spa100_ag.html

[[category:Analog Telephone Adapters]]

* You can check the most commonly used Star Codes from [[Cisco/Linksys Star Codes]]

=== SPA Star Codes ===

* *69 Call Return Code
This code calls the last caller.

* *07 Call Redial Code
Redials the last number called. (Not in pap2t)

* *98 Blind Transfer Code
Begins a blind transfer of the current call to the extension specified after the activation code.

* *66 Call Back Act Code
Starts a callback when the last outbound call is not busy.

* *86 Call Back Deact Code
Cancels a callback.

* *05 Call Back Busy Act Code
Starts a callback when the last outbound call is busy. (Not in pap2t)

* *72 Cfwd All Act Code
Forwards all calls to the extension specified after the activation code.

* *73 Cfwd All Deact Code
Cancels call forwarding of all calls.

* *90 Cfwd Busy Act Code
Forwards busy calls to the extension specified after the activation code.

* *91 Cfwd Busy Deact Code
Cancels call forwarding of busy calls.

* *92 Cfwd No Ans Act Code
Forwards no-answer calls to the extension specified after the activation code.

* *93 Cfwd No Ans Deact Code
Cancels call forwarding of no-answer calls.

* *63 Cfwd Last Act Code
Forwards the last inbound or outbound calls to the extension specified after the activation code.

* *83 Cfwd Last Deact Code
Cancels call forwarding of the last inbound or outbound calls.

* *60 Block Last Act Code
Blocks the last inbound call.

* *80 Block Last Deact Code
Cancels blocking of the last inbound call.

* *64 Accept Last Act Code
Accepts the last outbound call. It lets the call ring through when do not disturb or call forwarding of all calls are enabled.

* *84 Accept Last Deact Code
Cancels the code to accept the last outbound call.

* *56 CW Act Code
Enables call waiting on all calls.

* *57 CW Deact Code
Disables call waiting on all calls.

* *71 CW Per Call Act Code
Enables call waiting for the next call.

* *70 CW Per Call Deact Code
Disables call waiting for the next call.

* *67 Block CID Act Code
Blocks caller ID on all outbound calls.

* *68 Block CID Deact Code
Removes caller ID blocking on all outbound calls.

* *81 Block CID Per Call Act Code
Blocks caller ID on the next outbound call.

* *82 Block CID Per Call Deact Code
Removes caller ID blocking on the next inbound call.

* *77 Block ANC Act Code
Blocks all anonymous calls.

* *87 Block ANC Deact Code
Removes blocking of all anonymous calls.

* *78 DND Act Code
Enables the do not disturb feature.

* *79 DND Deact Code
Disables the do not disturb feature.

* *65 CID Act Code
Enables caller ID generation.

* *85 CID Deact Code
Disables caller ID generation.

* *25 CWCID Act Code
Enables call waiting, caller ID generation.

* *45 CWCID Deact Code
Disables call waiting, caller ID generation.

* *26 Dist Ring Act Code
Enables the distinctive ringing feature.

* *46 Dist Ring Deact Code
Disables the distinctive ringing feature. The default is *46.

* *74 Speed Dial Act Code
Assigns a speed dial number.

* *16 Secure All Call Act Code
Makes all outbound calls secure.

* *17 Secure No Call Act Code
Makes all outbound calls not secure.

* *18 Secure One Call Act Code
Makes the next outbound call secure. (It is redundant if all outbound calls are secure by default.)

* *19 Secure One Call Deact Code
Secure One Call Deact Code Makes the next outbound call not secure. (It is redundant if all outbound calls are not secure by default.)