[asterisk-users] Debugging DTMF

[Adrian Marsh]

All,

 

Ok, so I found out that my DTMF issues seem to be due to taking the
latest 1.4.19.1 release.

Originally I was on 1.4.18

I have a Cisco 7940, a Zoiper softphone and a Siemens IP phone.

I use two VoIP providers, Gradwell and Zen. Both over SIP

 

Heres what should happen:  I call 3rd party, get announcement, enter
DTMF, 3rd party plays it back.

 

Heres what I found:

 

Calling from the Cisco via 1.4.19.1 out to Gradwell, to the 3rd party
conference system, I would hear their welcome, and start entering DTMF.
The playback would start (so obviously its "hearing" something), but no
playback so it didn't hear the DTMF (I think).

 

Make the same call, but go via Zen instead, and all works ok (same
Asterisk, same Cisco)

Make the same call, but from the Zoiper or Siemens phone, via 1.4.19.1
and via Gradwell and all works ok.

 

Downgrade to 1.4.18 and I have no problem on any phone.

 

So it seems to be a Cisco 7940 vs 1.4.19.1 vs Gradwell problem.

 

I managed to find DTMF traces in an ethereal dump to Gradwell, so I know
that 1.4.19.1 is at least sending something to Gradwell.

 

I'm not sure what release of Asterisk that Gradwell will be on.

 

Any ideas?  I've downgraded my primary system back to 1.4.18 and will
upgrade the backup to 1.4.19.1 tomorrow to test a bit more.

 

Very confusing..

 

Adrian

 

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Adrian
Marsh
Sent: 29 April 2008 15:49
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Debugging DTMF

 

Hi All,

 

I'm trying to debug DTMF issues I have with certain endpoint
conferencing systems (external, 3rd party).

 

On our A*k server I log DTMF, and I see that coming through in the log.

What I'd like to see is what is sent onto our VoIP carrier over SIP.

 

I can do a tcpdump of the packets, but what am I then looking for?
Would it be in the RTP audio stream or within the SIP protocol??  I'm
using Wireshark to decode...

 

Thanks,

 

Adrian

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