Solved: SIP Inbound one way audio on transfers

balitewiczp · ‎01-29-2014

PSTN-->SIP-->CUBE-->>SIP-->CUCM. Outbound calls no problems at all.

Inbound calls completes, audio is good. But transfers (local IP-IP Phones)results is one way audio. IP phone cannot hear PSTN Caller. There are no sip bind commands, as Im running 12.4(22)T. I am defaulting to g729 , and Im thinking the transfers use g711, as the phones are in the same region. Could this be my problem.

dial-peer voice 1000 voip

description Internal CUCM dialing

destination-pattern 1...

session protocol sipv2

session target ipv4:10.0.15.40

dtmf-relay rtp-nte

ip qos dscp cs3 signaling

no vad

dial-peer voice 100 voip

description Inbound from SIP

translation-profile incoming 10_to_4_IN

session protocol sipv2

incoming called-number .

dtmf-relay rtp-nte

Manish Gogna · ‎01-29-2014

Try setting g711 codec for the region setting between sip trunk and ip phone as well. You can try it during off production hours and see if the issue goes away. Else, we can check the detailed ccm traces to see if the calling party is being updated about the final destination upon transfer.

HTH

Manish

View solution in original post

Manish Gogna · ‎01-29-2014

Can you try hardcoding g711 on the dialpeer and see if that fixes the issue.

Manish

balitewiczp · ‎01-29-2014

Sure I will try. But the Sip trunk is in a different region than the phones, the idea being to go 729 end to end. If i hardcode g711, what changes it to g729 for the phones? Or am I missing something.

Manish Gogna · ‎01-29-2014

Try setting g711 codec for the region setting between sip trunk and ip phone as well. You can try it during off production hours and see if the issue goes away. Else, we can check the detailed ccm traces to see if the calling party is being updated about the final destination upon transfer.

HTH

Manish

balitewiczp · ‎02-03-2014

I got pressed for time. I set the dial peer codec to 711, but couldnt figure out how to change the region relationship. If you could point me to a doc, that would be great. I ended up creating hardware mtp/transcoders, and placing it in the mrgl of the sip trunk.and this solved it. I thank you for your help

Manish Gogna · ‎02-03-2014

If setting up the transcoder fixed the issue then definitely it was a codec mismatch. You can change the codec settings through the 'Region' setting on the Device Pool assigned to a Phone / Gateway / SIP trunk and other devices.

HTH

Manish