SRST Voicemail Integration with an Analog/T1 CAS Line

If you need to integrate your SRST router with analog/T1 CAS link to Unity, then you nedd to adapt the configuration as follows:

!

call-manager-fallback

voicemail 12345

!

vm-integration

pattern direct * CGN

pattern ext-to-ext busy #FDN#2

pattern ext-to-ext no-answer #FDN#2

pattern trunk-to-ext busy #FDN#2

pattern trunk-to-ext no-answer #FDN#2

!

So you must use the configuration under the vm-integration command to enable voicemail integration with DTMF and analog mail systems. Here are a short explanation of all settings:

  • Pattern direct represents the fact that you press the message button
  • Pattern ext-to-ext is the forwarding to voicemail from an extension to another extension
  • Pattern trunk-to-ext represents the forwarding to voicemail from an external trunk to an extension

When configuring the pattern direct, pattern ext-to-ext and pattern trunk-to-ext, you must also specify combinations of alphanumeric strings (fewer than 4 digits in length) as well the calling number (CGN), called number (CDN) or forwarding number (FDN) to be sent to the voicemail system.

Then imagine the following scenario : Phone 1 ( DN 1001) is calling Phone 2 (DN 1002) which is set up to forward all busy and no-answer calls to the VM

We have then 1001 is the CGN, 1002 is the FDN and the voicemail system is the CDN. In other words, the calling number (CGN) is the number of the call originator, the frowarding number is the number of the extenson which is forwarding the call to the voicemail.So let’s take back our previous example and see what the configuration looks like:

!

pattern ext-to-ext busy #FDN#2

!

will represent to dial the number 123456789 (VM system) and send to the VM the following string in DTMF #1002#2 which must route the call correctly to the right user mailbox.

SRST on MGCP gateway

If we want to have SRST on MGCP gateways, ensure that you have the two following commands:

!
service alternate
call-manager fallback-mgcp
!

This will ensure that the MGCP gateway can provide Call Processing with SRST. Don’t forget also to have an H323 config to take over in SRST as MGCP hasn’t any Call Control.

SRST : Quick Definition

Survivable Remote Site Telephony (SRST) is a feature which ensures that IP Phones can continue to function even if they are unable to communicate with Call Manager. During a failure, Cisco IP Phones register with the local SRST router which provides Call Processing and Control.

With the Connection Monitor Duration, Cisco IP Phones do not fail back immediately to ensure that the Call Manager in question is back online and is stable.

Here is also another presentation of Cisco SRST ( Cisco video)

Unity Dialing Domain

A dialing domain is a group of Unity servers that share the same directory and do not require subscribers to use any type of prefix when performing transfers and sending messages between the servers (Extension must be unique within the dialing domain).

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