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Cisco Voice Gateway - SRST Configuration

RS19
Level 4
Level 4
I have the below configuration in my Cisco Voice Gateway.
Would like to understand what the below configuration means ?
What is the purpose of the below configuration ?

call-manager-fallback
  max-conferences 4 gain -6
  transfer-system full-consult
  ip source-address 10.10.10.1 port 2000
  max-ephones 50
  max-dn 300 dual-line preference 10

10.10.10.1 -> IP Address of the Voice Gateway
8 Replies 8



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So does the above configuration means that when CUCM is not reachable. the Cisco IP Phones will be registered with VG ?

Is my above understanding right ?

If my above understanding is right, will Cisco IP Phone extension - extension calling will work ?
What happens with the DID calling. Each IP Phone extension is given DID number ? Will DID calling works when CUCM is down ?

Yes, during CUCM outage phone register with vg. 
extension calling will work between the iP phones registered under the srst router.

 

if pri lines are terminated on same vg DID calling will work.

you can refer below guide and it explains more about both sccp and sip srst.

 

https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cusrst/admin/sccp_sip_srst/configuration/guide/SCCP_and_SIP_SRST_Admin_Guide/srst_overview.html



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Thanks,

But in the Call Manager fallback configuration, there is no configuration related to DID.
There is no information in the configuration which I shared related how DID calling will work in case of CUCM fails.

 

So u say even with the above configuration when CUCM is failed, when a call is received on DID it will be directed to the IP Phone extension.

When the system automatically detects a failure, Cisco Unified SRST uses Simple Network Auto Provisioning (SNAP) technology to auto-configure a branch office router to provide call processing for the Cisco Unified IP phones that are registered with the router.

 

When configuring   SRST fallback support using Cisco Unified CME you prebuilt phone and extensions configurations. so in your case you won't see and phone or DN configuration  on the router.

 

You might be  using some translations and we use similar translation to Make the calls working in SRST mode. @Roger Kallberg  has posted an example configuration of translation in srst. 

 

But for a scenario where all users share HQ PRI lines

HQ>> contain CUCM, and VG where PRI is terminated

Branch >> SRST router  with FXO lines.

In this case your DID calls will not work. 

 

 

 

we can confirm if you share your configuration.

 



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The phones and the gateway will exchange information to dynamically configure the phones and directory numbers for when they go into SRST mode. As such there is no actual configuration for directory numbers or devices on the gateway. If the directory numbers in CM are not matching the received numbers as sent from the telco you likely have some translations/transformations in-place already. Depending on where you do this you might need to modify the configuration in the gateway and then it would be as @Nithin Eluvathingal wrote, you would use voice translation rules to change the called number received by the telco to match the directory numbers you have set in CM as they will by magic show up in the gateway.



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Scott Leport
Level 7
Level 7

Hi there,

 

Voice Translation rules when applied appropriately usually take care of this and are not present in the SRST configuration.

Alternatively, there is the dialplan pattern command which can be used under the Call Manager fallback configuration, e.g.

 

call-manager-fallback
dialplan-pattern 1 4085354... extension-length 4

 

This example would truncate a 10 digit number down to 4 and could be used on incoming calls from PSTN.

 

 

Do not use dialplan-pattern configuration, it's an old outdated legacy way of configuring things. You can use voice translations rules in SRST, below is an example on how to get phones in SRST to use translations. Besides from this you would also for sure need to modify the called and also likely calling numbers on egress into the gateway from the telco, but as that would also likely be the case for the normal call state scenario that could already be in place. If you do have a situation where you don't do the number manipulations ingress to the gateways from the telco, let's say you do it in the CM on ingress on the trunk, I would recommend that you change this to be done in the gateway instead as then you'd only have one place where this needs to be maintained instead of multiple.

voice translation-rule 50
 rule 1 /^4662010$/ /000995552010/
 rule 2 /^4665555$/ /000995555555/
 rule 3 /^892\(8[089]..\)$/ /+89777917\1/
 rule 4 /^892\(82..\)$/ /+089777917\1/
 rule 5 /^\+8977\(.*\)/ /0\1/
 rule 6 /^\+89\(8.*\)/ /00\1/
 rule 7 /^\+89\(.*\)/ /00\1/
!
voice translation-profile SRST-IN
 translate called 50
! call-manager-fallback translation-profile incoming SRST-IN

 



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