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

[Brian Meade]

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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>
>
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