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SIP Trunk Failover to an h323 gw timer

Hi

I have just configured a couple of Sip Trunks to a couple of GSM GWs for cel phone calls.

My Route pattern is 9044.XXXXXXXXXX and discard digits predot. Because the celular SIMS have been setup to only receive 10 digits.

RL   1st   RG_SIP_Trunk

       2nd  RG_H323          Prefix 9044

When we dial any cell phone number the call goes to the sip trunk and reaches the GSM gateway, when all SIMS are busy, calls go to the h323 gateway and calls are completed.

It's working as desired, but when I disconnect the GSM gws from the network and they are not reacheable the call takes around 70 seconds to failover to the h323 GW.

We dial 9044 5513848588 for a local cel phone.

I have been playing sith the SIP profile timers but I does not change anything.

Regards.

3 Replies 3

Jonathan Schulenberg
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You must have more than one SIP trunk and/or multiple GSM gateways? I ask because 70 seconds is way longer than it would take for a single SIP trunk to fail if there is no response to the INVITE using default settings. By default the SIP trunk should fail after 3.5 seconds (1 INVITE + 6 INVITE Retries * 500ms). A good way to bring this down would be to change the INVITE Retries from six (6) to two (2) on the SIP Profile. You could also use a tool such as TranslatorX to parse Detail-level SDI logs from CUCM and see what's happening in the dialog.

Otherwise you could see if the GSM gateways support SIP OPTIONS PING (requires CUCM 8.5+). This is a keepalive mechanism that causes CUCM to consider the SIP trunk down if in-dialog SIP messages do not receive a response.

Please remember to rate helpful responses and identify helpful or correct answers.

Hi Jonathan thanks for the reply

I found this document

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_configuration_example09186a008082d76a.shtml

And it states that the first retry starts with 500ms and then it doubles the time for each new retry, at least that is what I understood. I will try reducing the retries and using the tool you mentioned.

Yeah I read about the SIP Options Ping unfortunately this is a CUCM 8.0.3.

Found the timer at service parameters and worked

for me.