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" <bmead...@vt.edu>
Inviato: ‎02/‎03/‎2015 21:57
A: "Alessandro Bertacco" <bertacco.alessan...@alice.it>
Cc: "cisco-voip@puck.nether.net" <cisco-voip@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.alessan...@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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