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SIP Phones don't re-register on CUCM after SRST finished


I have a 2901 router as MGCP gateway on the CUCM.

When link between router and CUCM fails, SCCP and 3rd party SIP phones register correctly on the 2901.

But when SRST finishes and network returns in normal conditions, SIP phones keep registered on 2901 whereas SCCP phones re-register on CUCM.

I think 2901 doesn't aware that SIP Proxy CUCM is online, it may be a keepalive command to add or something but I've not found.

Here is a part of my conf :

Subscriber IP :

Publisher IP :

2901 IP :

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

no supplementary-service sip handle-replaces

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none


  registrar server


voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw

codec preference 3 g726r32



voice register global

system message /!\ Mode de secours actif /!\

max-dn 4

max-pool 4


voice register pool  1

id network mask

preference 2

proxy preference 1 monitor probe icmp-ping

voice-class codec 1


ccm-manager fallback-mgcp

ccm-manager redundant-host

ccm-manager mgcp

no ccm-manager fax protocol cisco

ccm-manager music-on-hold

ccm-manager config server

ccm-manager config



mgcp call-agent 2427 service-type mgcp version 0.1

mgcp dtmf-relay voip codec all mode out-of-band

mgcp rtp unreachable timeout 1000 action notify

mgcp modem passthrough voip mode nse

mgcp package-capability rtp-package

mgcp package-capability sst-package

mgcp package-capability pre-package

no mgcp package-capability res-package

no mgcp package-capability fxr-package

no mgcp timer receive-rtcp

mgcp sdp simple

mgcp fax t38 inhibit

mgcp rtp payload-type g726r16 static


mgcp profile default


sccp local GigabitEthernet0/0

sccp ccm identifier 10 version 7.0

sccp ccm identifier 20 version 7.0



sccp ccm group 1

associate ccm 20 priority 1

associate ccm 10 priority 2

switchback method graceful



max-conferences 2 gain -6

transfer-system full-consult

user-locale FR

ip source-address port 2000

max-ephones 30

max-dn 30

system message primary /!\ Mode de secours actif /!\

system message secondary Mode degrade actif

no huntstop

alias 1 7440 to 8888 preference 3

time-format 24

date-format dd-mm-yy


Thank for your help !



Rising star

Hi Clement,

When phones fallback to SRST mode is because have been 3 missed TCP Keepalive between phones and CUCM. When wan connection is retored; phone re-established TCP connection with CUCM so that's when they re-register with CUCM. So the router never send a message telling the phone they need to go back to re-gister with CUCM.

What phone model you're working with?

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It's Polycom DECT 3rd party SIP phones.

These DECT are connected to a Polycom KWS8000 Wireless Server, and this is the KWS8000 which convert streams to IP and establish SIP sessions with CUCM.

All proxies are configured in the KWS8000, so when subscriber is not available, it tests the second proxy, and finish by SRST.

But when subscriber becomes available, it keeps connection to 2901 because link with it is already OK, so no reason to change.

I think it was 2901 job to inform phones that original proxy is now avalaible as it is not in SRST mode anymore ?


The 2901 should not inform the phones to register back to the CUCM.

From the SRST System Administrator Guide:

In SIP phone load 8.0 and later versions, SIP phones use keepalive to maintain a connection with Cisco Unified SRST during active registration with Cisco Unified Communications Manager. Every two minutes, a SIP phone sends a keepalive message to Cisco Unified SRST. Cisco Unified SRST responds to this keepalive with a 404 message. This process repeats until fallback to Cisco Unified SRST occurs. After fallback, SIP phones send a keepalive message every two minutes to Cisco Unified Communications Manager while the phones are registered with Cisco Unified SRST.


Did you ever find a solution to your issue? I'm running into the same issue with Cisco 7962s that are running SIP.

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