Supervised transfer DTMF problem

[dlynton]

I'm using Microsoft Speech Server to initiate a supervised transfer. After connecting the two callers, neither party can hear DTMF tones that they punch in. The tones come through sounding like a click. This doesn't happen when I do a blind transfer. Can you speculate if this is being caused by PBXNSIP and if there is a way to troubleshoot it?

 

Thanks,

Daniel

[Vodia PBX]

Well, that is a complicated topic. DTMF depends not only on the connection between the PBX and Exchange, is also depends on the DTMF that has been negotiated on the other side. In the end, a Wireshark trace will show what is the problem.

 

Plus it is still a mystery to me how Exchange can be made to detect inband DTMF. They are able to recognize voice, so they should also be able to detect inband DTMF IMHO. I saw it working in one place, but in other places it did not work and I have no idea why.

[dlynton]

When performing a wireshark trace there isn't any traffic when I send DTMF tones. The trace at the time of the transfer is almost identical. After this point the call should be only between the caller and the called party. I see a SIP REFER, and then a BYE response on both the supervised and blind transfer calls.

 

Here is the only difference I've found between the blind transfer (DTMF can be heard) and the supervised transfer (DTMF tones sound like clicks).

 

Blind transfer: REFER-TO: <sip:7132234676@192.168.54.103:5060;transport=tcp;user=phone>

 

Supervised: REFER-TO: <sip:7132234676@192.168.54.103:5060;transport=tcp;user=phone?REPLACES=2d886047-461f-483c-b59e-179845662721%3Bto-tag%3D7b97da7e0c%3Bfrom-tag%3D264b596bbf>

 

Do you have any idea what the REPLACES querystring is doing?

 

Thanks for your help! We are so close to being able to deploy our new IVR application on MS Speech Server + PBXNSIP! This is our last issue after a barrage of problems.

[Vodia PBX]

Well, the Replaces header tells the "other side" what call to drop as a replacement for the new call (initiated by the REFER). Because everything is happening inside the PBX, that logic is programmed without really initiating another call. Doe the replaced Call-ID exist on the PBX? If not, the PBX should log the message "Replaces: Call-ID xxx not found" on log level 5.

[dlynton]

I searched for replace in my log and found the sip header as well as pbxnsip reporting "Supported: 100rel, replaces, norefersub". I did *not* see any instance of "Replaces: Call-ID...not found". I see PBXNSIP assign a call id of ced43e0d-b8a6-46b1-bfb0-3da0b8e9d40a and then that is used later in the REPLACES= of the sip header. Do you know what the to-tag and from-tag are for? I'm attaching the log from my call (I have the log level set to 10--what can I set it to while being confident I'll get all the relevant messages?). The SIP REFER looks like it's doing everything right, and the supervised transfer works great except for the DTMF tones. Any other ideas?

log.txt

[Vodia PBX]

I don't think it is the Replaces. I looked at the SDP, but they all contain the out of band codec. It is wierd.

[dlynton]

Is there a codec preference or anything else I can try changing in the trunk settings? Anything else you can recommend to troublshoot this problem? I assume that a supervised transfer / conference call would typically work fine in your pbx? i.e. DTMF tones are usually heard by all parties?

 

If you recall, I'm the one you helped set up for connectivity with Speech Server & Asterisk. Do you know anything about Asterisk that we should be checking to make sure the DTMF gets passed through? I noticed in the Asterisk call log "getdtmf on channel 41: Operation now in progress" only appeard once during the two test calls I made.

 

It's quite frustrating to be so close, yet so far! We've come a long way to get everything working between MSS, Asterisk and PBXNSIP. It will be such a relief to finally get this behind us, so any other help you can offer is greatly appreciated!

[Vodia PBX]

I think we need a Wireshark trace here...