SRST Voice Gateway Configuration

[Hary_CsC]

Hai Guys

I just installed Voice Gateway Router with the version 17.5 (SLUP), but I haven't registered to the CSSM Server.

Is this the factor why my VG Router can't SRST ?

Here's the SRST configuration that i configured on VG.

Please Advise

Thank You

[b.winter]

Are you using SRST on a CUBE with a SIP trunk to the provider?

If yes, it is mandatory to use a tenant configuration

https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-cube-multi-tenants.html

There were also some other posts about tenants the last weeks, you might also wanna check them out

What you are definitely missing:

voice service voip
 sip
  registrar server expires max 240 min 60
  bind all source-interface <internal interface>
!
voice register global
 default mode (is srst)
 dialplan-pattern 1 .... (4-digit-dn)
!
call-manager-fallback
 dialplan-pattern 1 .... (4-digit-dn)
!
no sip-ua

--- Please rate this post as "Helpful" or accept as a solution, if your question has been answered ---

[Roger Kallberg]

For reference this is from our configuration template for voice gateways.

[A.A.A.A] = LAN Interface IP Address (Voice Vlan)  

SRST configuration

The below is functions in SRST configuration that are shared between SCCP and SCCP.

•	MOH configuration, file(s) and multicast.
•	Alias configuration.
•	Default destination.
•	Date and time format.
•	Conference settings.
•	Transfer settings.
•	Secondary dial tone.

voice translation-rule 50
 rule 1 /^466xxxx$/ /00046xxxxxxxx/
 rule 2 /^466xxxx$/ /00046xxxxxxxx/
 rule 3 /^632\(....\)$/ /+632xxx\1/
 rule 4 /^\+632\(.*\)/ /0\1/
 rule 5 /^\+63\(.*\)/ /00\1/
 rule 6 /^\+\(.*\)/ /000\1/
!
voice translation-profile SRST-IN
 translate called 50


SCCP SRST configuration

ccm-manager music-on-hold

call-manager-fallback
 translation-profile incoming SRST-IN
 secondary-dialtone 0
 max-conferences 8 gain -6
 transfer-system full-consult
 timeouts interdigit 5
 ip source-address [A.A.A.A] port 2000
 max-ephones 128 ;set max-ephones to the maximum that router model can support
 max-dn 256 dual ;max-dn should be set to double amount of max-ephones
 system message primary SRST Failover
 default-destination +632xxxxxx
 moh SampleMOH.wav
 multicast moh 239.10.138.16 port 16384 route [A.A.A.A] 192.168.255.10
 time-zone 41
 time-format 24
 date-format dd-mm-yy


Note: If the site has FXS ports that are controlled via SCCP this is needed for them to operate in SRST as they will not use the normal SRST reference setup in CUCM.

sccp ccm [A.A.A.A] identifier 1 version 7.0 
!
sccp ccm group 10
  associate ccm 1 priority 4


SIP SRST configuration

voice service voip
 allow-connections sip to sip
 sip
  registrar server
!
voice register global
 default mode
 system message SRST Failover
 max-dn 256 ;max-dn should be set to double amount of max-pool
 max-pool 128 ;set max-pool to the maximum that router model can support
 timezone 41
 phone-mode phone-only
!
voice register pool 1
 id network x.x.x.x mask y.y.y.y	;x.x.x.x,/y.y.y.y refer to network range assigned to entire site, not just the voice vlan
 translation-profile incoming SRST-IN
 dtmf-relay rtp-nte
 voice-class codec 1
 no vad
!
sip-ua
 registrar ipv4:[A.A.A.A] expires 3600

Hopefully that could be of help to you.

[Hary_CsC]

Hai @Roger Kallberg

Sorry for the late reply.

Whether the above configuration is different if using SIP between SIP?

Thank You

[Roger Kallberg]

What do you mean by “using SIP between SIP”?

[Hary_CsC]

Hai @Roger Kallberg 

I have an IP Phone with SIP Protocol, can this configuration also be used ?

Thank You

[Roger Kallberg · Accepted solution]

Word of notice, I have not looked at the last configuration, or whatever you shared.

What I shared earlier is for both SIP and SCCP SRST. This should cover what you need.