No DTMF tones on the SIP trunk until CUC delivers call to VM

[jmhunter]

Hey Folks,

Seems I have an issue where my DTMF tone is either not recognized or allowed when reaching my Cisco Unity Connection (CUC) server.

What I am attempting to do is hit * when I dial into my VM system to get to a login prompt to check my voicemail, all from an outside line (PSTN).

Here is a previous thread I went through trouble-shooting this and seem to determine the issue may be with the SIP trunk(s) we use.

https://supportforums.cisco.com/thread/2108830?tstart=0

If I dial the CUC server internally, this works fine.  I dial from the PSTN, the DTMF tones are ignored.  The above link will show the steps taken to make sure "Ignore Caller Input" is not checked, etc.

Any ideas

Thanks,

James

[Carlo Poggiarelli]

Hi James.

How did you configured your Voice Gateway (MGCP,H323,SIP)?

It would be usefull if you can post your VG Config so we can check any misconfig...

Please let us know

Regards

Carlo

[jmhunter]

Thanks Carlos. I was afraid that no one would be able to reply to this. I've attached the config (And some of it is a mess).  Like so often heard elsewhere, I indeed inherited this mess, so there is a bit of ramp up (not to mention clean up) process for me to go through. Thanks...

James

[Carlo Poggiarelli]

HI James.

Since you are using h323 dial-peers try to use dtmf-relay h245-alphanumeric otherwise try to add "session protocol sipv2" pointing to CUCM.

HTH

By

Carlo

[jmhunter]

Thanks Carlos.  Since I have little experience with Cisco Voice (None with SIP implementations), at what point in the config file will I apply these parameters, and what impact when I issue the command(s) will any users see.

Of course, I can immediately back out the changes, (add the NO in front of the line to remove), and plan on doing this early morning before most users even show up.

From looking up that command(dtmf-relay h245-alphanumeric), I would apply that change to one of the dial-peer voice XX groups, or would I apply that to all of those groups?

And as for the second command you mentioned (session protocol sipv2), does this go in the dial-peer voice XX group as well?

Thanks,

James

[cjoseph23]

James,

I had a similiar issue with DTMF over SIP to Unity (not connection).  It would work internally but not from the SIP Provider.  I got it workgin with the following Dial peer config.  This is a lab enviroment so i was able to play with the settings. 

Also i am using g711 end-to-end.  I am not trancoding the calls to g729 and the provider is sending the calls to me g711.

dial-peer voice 3 voip

description INBOUND XXX SIP to CUCM

destination-pattern XXX.......

voice-class codec 1

session protocol sipv2

session target ipv4:xxx.xxx.xxx.xxx

dtmf-relay rtp-nte

no vad

CJ

[jmhunter]

Thanks CJ.  No test lab here (small company).

If I am reading this correctly, I would probably apply the session protocol sipv2 to my Dial-peer voice 1 voip, 3, 4, & 21451. dial-peer voice 2 already has this command applied.

I don't know.  I seem to be lost in this mess.  Guess I could open a TAC case.