Call Forward All to Voicemail not working using SIP

[Aaron Ulysses Lorenzo]

Hi Guys,

I am having problem with my inhouse project and I need your assistance. The problem is CFA (Call Forward All) to Voicemail gives me one ring then busy tone. It looks like codec to me but I cannot figure out how to execute it. Here are my call scenarios that might help.

CFA Scenario 1

- Incoming call to SIP Number then forwarded to another SIP Number - OK/Passed

- Callers are from both internal and external

CFA Scenario 2

- Incoming call to SIP Number then forwarded to Voicemail - OK/Passed

- Caller is from internal party

CFA Scenario 3  Problem !!!!

- Incoming call to SIP Number then forwarded to Voicemail - Failed (one ring then fast busy tone)

- Caller is from external party

Caller External (mobile/fixed #) -----> PSTN ----> SIP - SIP - CUBE--->CUCM 7 ---> Phone ---->Voicemail

I can see from the debug that the call is routed up to the voicemail pilot but I reality I did not hear the voice announcement. Then after few seconds I receive lots of email from the exchange server for the voicemail. I am reaching the voicemail but it does not stay there and allow me to leave message.

Other isolation done was conference call and it worked. We have transcoder configured and even tried changing the regions as I have read from the previous discussion but unfortunately did not help.

Any ideas would be appreciated...

[Ayodeji Okanlawon]

Can you send the debug ccsip messages. Include calling and called number. Also send your sh run


Sent from Cisco Technical Support Android App

[Ayodeji Okanlawon]

Can you send us the ff:

Debug ccsip messages
Sh run
Include called and calling number for the debugs


Sent from Cisco Technical Support Android App

[Aaron Ulysses Lorenzo]

Hi,

Below are the debug ccsip messages and sh run as requested. In the debug, there is only one call attempt and it looks like it is looping. I said looping because I receive multiple email notifications from the exchange even before the call is terminated.

Thank you very much in advance!

[Ayodeji Okanlawon]

AAron,

What is the region setting between the sip trunk in cucm and the Destination 9712715364. You have configured your dial-peer to use G711 to CUCM. It looks as though the region setting between the phone and the siptrunk is set to G729. Even though you have a xcoder configured, is the region between the CUBE and the xcoder set to G711?

You can do a quick test by configuring either voice class codec with g729 as the preferred and then apply it to the dial-peer 666 or by setting the codec on your dial-peer "dial-peer voice 6666 voip" to use G729

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[Aaron Ulysses Lorenzo]

@ aolkanlawon,

I did your suggested quick test to force dial-peer 6666 to use codec G729 but I got the same result. Sorry but how do I get the region relationship between CUBE sip trunk and VM sip trunk,?

[islam.kamal]

dear

based on the above debug .I found that you have the Callmanager sent 503 service unavailable with cause value 47, "Resource unavailable". Looks like CCM is trying to enagage transcoder. Detailed CCM logs will reveal that. But can you check the  region relationship between CUBE sip trunk and VM sip trunk, and make sure to use codec g729 instead of g711alaw which  is negotiated by CUBE to CCM.Kindly find the below link

http://www.cisco.com/en/US/docs/ios/12_2t/12_2t11/feature/guide/ftmap.html

Thank you

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