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ATA190 on CUCM10.5.2 don't revert to CUCM when go in SRST fallback

Hi,

  anyone have this problem?

 

I've One ATA190 (1.1.2 (005) Feb 6 2015) registered with CUCM 10.5.2, that when goes to SRST fallback, and then when Primary connection CUCM is up again, don't reconnect to CUCM but remain connected to the SIP SRST service.

All other phones revert to CUCM automatically.

 

thank you

 

Rergards

 

Alessandro Bertacco

4 Replies 4

Mohammed Khan
Cisco Employee
Cisco Employee

Hi Alessandro,

Interesting let me test this in lab. can you provide ata xml config file and "show tech voice" from the gateway.

Regards,

Mohammed Noor

Hi Mohammed and thank you very much for your answer.

Differently as I've said above, all SIP phone are impacted from my issue.

 

So, when connection to the CUCM is broken, all SIP phone correctly register on the SRST gateway version 15.4(3)M2, but when communication with CUCM go up again no one phone reconnect to the CUCM. To force that I need to remove "voice register global" command from the router. After about 2 minutes phones come back to the CUCM.

Only SIP phone are affected from this, phone used in my environment are:

7821 (sip78xx.10-2-1-12SR1-4)

9951 (sip9951.9-4-2-13)

ATA190 (1.1.2 (005) Feb 6 2015)

 

Configuration of the SRST gateway are:

!
voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 supplementary-service h450.12
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 modem passthrough nse codec g711ulaw
 h323
 sip
  bind control source-interface GigabitEthernet0/1.20
  bind media source-interface GigabitEthernet0/1.20
  registrar server expires max 600 min 60
  no silent-discard untrusted

!

!
voice register global
 mode srst
 timeouts interdigit 7
 system message Systema SRST Attivo
 max-dn 116
 max-pool 58
!
voice register pool  1
 registration-timer max 120 min 60
 id network 192.168.101.0 mask 255.255.255.0
 dtmf-relay rtp-nte sip-notify
 voice-class codec 1
!

sip-ua

 

Hi bertaale,

Go ahead open a Cisco TAC case to investigate it.

Regards,

Mohammed Noor

Hello, did you find an solution for this? I have the same problem. regards Achim