No audio when call goes to the call handler

[JH51928]

We have two sip trunks from ITSP. One of the sip trunk is working fine, no issues. The other one is having issues when I forward the call to the handler, I hear nothing, no ringing, no greetings, and no audio at all. I tried putting dn but didn't work. It is fine if I put the dn on the translation pattern 

Call Flow

ITSP-siptrunk-CUBE->CUCM->CUC

I can see that it hit the dial-peer on the CUBE, with dtmf-relay rtp-nte sip-kpml sip-notify.

I'm seeing issues about DTMF, 

isMTPNeededForDTMF, No DTMF is possible for call since one (or both) party does not support any DTMF cap party1DTMF(3) party2DTMF(0)

[Scott Leport]

Hi there,

Are both SIP trunks on the same device or different device?

Do your outbound dial-peers towards CUCM have the same configuration, such as codec and DTMF relay settings?

Assuming a SIP trunk between CUCM and CUBE, do you have the same DTMF configuration on the CUCM SIP trunk settings?

What's your integration between CUCM and CUC? SIP or SCCP based?

Would you also be able to supply a debug ccsip messages of a call to Unity over the "working" SIP trunk and one over the SIP trunk where you have the issue please?

[JH51928]

Thanks for replying Scott,

SIP trunks are on different devices.

Both of the outbound dial-peers have the same configuration (codec and dtmf-relay settings)

Both CUBE have the same DTMF signalling setting (RFC 2833)

CUCM and CUC are integrated using SIP

- I have log for debug ccsip mess for the siptrunk that has issue

- I'll send the log for the working sip trunk later, having an issue with the syslog server it's not showing the correct output.

We use Translation Pattern for DID and the DN that I put in Called Party Transform Mask is the one I'm using for CTI route point.

I can see in cucm logs that it hit the VM pilot number

Also, I think CUCM and CUC are talking to each other.

[Anthony W. · Accepted solution]

Looking at your traces:

Incoming Dial-peer=1

Outgoing Dial-peer=11

Can you please post the config for those 2 dial-peers? Also what is the MRGL on each trunk and does it contain a MTP? If a MTP is trying to be invoked, your having a DTMF mismatch and we should be able to see that in on the dial-peer config. I am guessing you may not have a MRGL or an incorrect MRGL on one of the trunks.

For the audio issues, ensure you have your dial peers have the correct source interface:

voice-class sip bind control source-interface GigabitEthernet0/0/X
voice-class sip bind media source-interface GigabitEthernet0/0/X

[JH51928]

Using different source interface for the sip bind on the incoming dial peer fixed the issue.

I was able to get assistance from TAC, they said the loopback interface was not working (The loopback interface has the ip address of the cube). I forgot to ask why : ( 

Any ideas? 

[Anthony W.]

I would have to look at the full config but if you did not have the bindings set on the dial-peer, CUBE will look at the voice service voip section next. If nothing is set anywhere, CUBE tries to find the right interface (I would have to look up the selection criteria as I don’t know it off the top of my head) but doesn't always work correctly. Its best to set the source interface so there is no confusion on which interface the traffic is coming from. Your symptoms are exactly what you would expect when using the wrong source interface when it comes to audio. Your DTMF issue was something else.

[M Reza]

hi friends,

i had this issue too..

after "Check" MTP option (Media Termination Point) on SipTrunk between CUCM and Router-Cube, audio is working properly and fine..

my scenario was as this:

E1-Line-Telco  <<>>  CUCM  <<>> CUC

no audio if call goes to CUC, then dial any extension..