[cisco-voip] R: SIp Phones on CUCM10.5.2 don't revert to CUCM whengo in SRST fallback

[Alessandro Bertacco]

Hi Brian, thank you for the answer.
But if keepalive don't work, phones will unregister automaticalli, and it is not my case.

How I can check keepalive from phones?

In connection monitor duration under device pool I've leaved blank, I use the default value under Enterprise parameter that is set to 120.

Do you think is better to specify the value also under device pool?

Thank you very much.

Regards

Alessandro.

----- Messaggio originale -----
Da: "Brian Meade" <bmeade90 at vt.edu>
Inviato: ‎02/‎03/‎2015 21:57
A: "Alessandro Bertacco" <bertacco.alessandro at alice.it>
Cc: "cisco-voip at puck.nether.net" <cisco-voip at puck.nether.net>
Oggetto: Re: [cisco-voip] SIp Phones on CUCM10.5.2 don't revert to CUCM whengo in SRST fallback

Here's what you're probably hitting for the 7800 series- https://tools.cisco.com/bugsearch/bug/CSCus18070


For the other models, do you see the phones having successful keepalives with CUCM before you disable voice register global?


What do you have set for "Connection Monitor Duration" on the device pool?


On Mon, Mar 2, 2015 at 3:03 PM, Alessandro Bertacco <bertacco.alessandro at alice.it> wrote:

 
Hi All,
I’ve this problem:
 
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 at the Branch Site, 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, and wait about 2 minutes phones come back to the CUCM.
Only SIP phone are affected from this issue. 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)
No Keepalive issue, because phone after removing voice register global connect again to the CUCM. 
 
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
!
Can you help me?
 
Thank you regards
 
Alessandro Bertacco

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