Internal calls are not working in SRST mode ?

[vistaprogrammer]

Dear All :

thank you very much for this fantastic Support Community

i have the following case: two sites : headquarter with CCUM and branch site with voice gateway cisco router 3845 configured as SSRT mode as following :

- on call manager(10.50.3.11) the SRST congiguration is done to refer to the branch voice gateway

10.51.3.15 port:2000

- on the branch voice gateway(10.51.3.15), the configuration is :

dial-peer voice 10 voip
 description ---- Accept & Send CCM2
 destination-pattern 1.....
 session target ipv4:10.50.3.12
 voice-class codec 1  
 voice-class h323 1
 dtmf-relay h245-alphanumeric
 no vad

dial-peer voice 11 voip
 description ---- Accept & Send CCM1
 preference 1
 destination-pattern 1.....
 session target ipv4:10.50.3.11
 voice-class codec 1  
 voice-class h323 1
 dtmf-relay h245-alphanumeric
 no vad
!

!
application
 global
  service alternate default

call-manager-fallback
 secondary-dialtone 9
 max-conferences 4 gain -6
 transfer-system full-consult
 ip source-address 10.51.3.15 port 2000
 max-ephones 80
 max-dn 90
 application sccp_application
 system message primary System on Backup Mode
 dialplan-pattern 1 1..... extension-length 5
 transfer-pattern ......
 keepalive 20
 pickup 222
 multicast moh 239.1.1.1 port 16384 route 10.51.3.15
 time-format 24
 date-format dd-mm-yy
!

i am using the dial plan 1.....  and when the two WAN links are down between the two sites the network configuration on each cisco ip phone in the branch shows SRST as third call manager but the ip phones inside that branch can't call each other ?did i miss any step ,

i think my dial peer(10,11) is not working on the SRST mode since they are still referring to the call manager

please help

[Chris Deren]

You only have dial peers to CUCM, and I don't see any PSTN dial peers. Can you post the entire "show run" so we can determine which protocol you use MGCP vs SIP vs H323 and type of PSTN trunks you have?

[vistaprogrammer]

Thank you for replay , attached the show run command result...

please note that this configuration for the branch site voice gateway router

i am using h323, and i want the internal calls between that branch users to call each other ...

thank you

[Chris Deren]

Looks like you have many POTS dial peers pointing to PSTN. Replicate the issue and provide "debug voice ccapi inout" and "debug voice dialpeer"

[vistaprogrammer]

Dear Chris:

when fallback ,i don't want anything to go over PSTN ,

i want just the internal ip phones in the branch to call each other in that branch ...

hope that understood

[Chris Deren]

You could use after-hours block pattern feature with 24x7 option:

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/command/reference/cme_cr/cme_a1ht.html#wp3719778986