SRST Voicemail Integration with an Analog/T1 CAS Line

by | Mar 7, 2010 | H.323, IOS, Lab, Non classé, SRST, Written Theory | 2 comments

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 Voicemail Integration

by | Mar 7, 2010 | CUC, H.323, IOS, Lab, Non classé, SRST | 0 comments

Under SRST, you can still benefit of the voicemail system integration at the central site but this time instead to have an internal call to Unity, you need to make an outside calls to Unity when SRST is enbale.

So under the call-manager-fallback command, enable the voicemail <<VM_number_to_dial>> and your end-users will still have the opportunity to press the message button in SRST

SRST on MGCP gateway

by | Mar 7, 2010 | H.323, IOS, Lab, MGCP, Non classé, SRST, Written Theory | 1 comment

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 Basic Configuration

by | Mar 7, 2010 | IOS, Lab, Non classé, SRST | 0 comments

Here is the basic configuration in order to enable SRST:

!
call-manager-fallback
ip source-address 10.1.1.1 port 2000 strict-match
max-ephones 20
max-dn 40
!

Don’t forget also to put the SRST Reference under the Call Manager Device Pool

SRST : Quick Definition

by | Mar 7, 2010 | CUCM, Non classé, SRST, Written Theory | 0 comments

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)

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