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[asterisk-users] SIP DTMF Troubleshoot


 
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joakimsen at gmail.com
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PostPosted: Mon Jan 28, 2008 6:05 pm    Post subject: [asterisk-users] SIP DTMF Troubleshoot Reply with quote

How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no
messages related to DTMF... or if I just do a global SIP debug for
that matter.... I am using RFC DTMF but it's not being passed to the
PSTN and I need to debug this further. I've tried to increase the
verbosity and the debug ('set debug n') and that didn't help either. I
assume this is because even RFC2833 sends the DTMF as RTP which
wouldn't show up anyways.... but how to troubleshoot DTMF issues?
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jsmith at digium.com
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PostPosted: Mon Jan 28, 2008 6:47 pm    Post subject: [asterisk-users] SIP DTMF Troubleshoot Reply with quote

On Mon, 2008-01-28 at 18:05 -0500, Andrew Joakimsen wrote:
Quote:
How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no
messages related to DTMF... or if I just do a global SIP debug for
that matter.... I am using RFC DTMF but it's not being passed to the
PSTN and I need to debug this further. I've tried to increase the
verbosity and the debug ('set debug n') and that didn't help either. I
assume this is because even RFC2833 sends the DTMF as RTP which
wouldn't show up anyways.... but how to troubleshoot DTMF issues?

I'd first turn on "rtp debug" and see if that helps. If that's not
enough information, I'd go into logger.conf and add "dtmf" to the logger
and messages lines (and any others you care about), and then do a
"logger reload" from the Asterisk CLI.

--
Jared Smith
Community Relations Manager
Digium, Inc.
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abalashov at evaristes...
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PostPosted: Mon Jan 28, 2008 7:03 pm    Post subject: [asterisk-users] SIP DTMF Troubleshoot Reply with quote

I think your best bet is to do a packet capture and look for RTP packets
with an RTP Event payload ("rtpevent" display filter).

On Mon, 28 Jan 2008, Andrew Joakimsen wrote:

Quote:
How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no
messages related to DTMF... or if I just do a global SIP debug for
that matter.... I am using RFC DTMF but it's not being passed to the
PSTN and I need to debug this further. I've tried to increase the
verbosity and the debug ('set debug n') and that didn't help either. I
assume this is because even RFC2833 sends the DTMF as RTP which
wouldn't show up anyways.... but how to troubleshoot DTMF issues?

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--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : +1-678-954-0670
Direct : +1-678-954-0671
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joakimsen at gmail.com
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PostPosted: Mon Jan 28, 2008 7:17 pm    Post subject: [asterisk-users] SIP DTMF Troubleshoot Reply with quote

Too much info then too little info.

Basically the issue is the provider this happens even when we send
them the calls in IAX because they talk SIP to the same gateway.

I just need to prove it to these people. Anyone have any DTMF issues
between Asterisk and a Quintum gateway?

On Jan 28, 2008 6:47 PM, Jared Smith <jsmith at digium.com> wrote:
Quote:

On Mon, 2008-01-28 at 18:05 -0500, Andrew Joakimsen wrote:
Quote:
How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no
messages related to DTMF... or if I just do a global SIP debug for
that matter.... I am using RFC DTMF but it's not being passed to the
PSTN and I need to debug this further. I've tried to increase the
verbosity and the debug ('set debug n') and that didn't help either. I
assume this is because even RFC2833 sends the DTMF as RTP which
wouldn't show up anyways.... but how to troubleshoot DTMF issues?

I'd first turn on "rtp debug" and see if that helps. If that's not
enough information, I'd go into logger.conf and add "dtmf" to the logger
and messages lines (and any others you care about), and then do a
"logger reload" from the Asterisk CLI.

--
Jared Smith
Community Relations Manager
Digium, Inc.


_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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joakimsen at gmail.com
Guest





PostPosted: Tue Jan 29, 2008 10:30 pm    Post subject: [asterisk-users] SIP DTMF Troubleshoot Reply with quote

Everything seems find on my end. Here's the setup:

Linksys SPA922 <-----> Asterisk 1.4 <-------> Quintum T1 gateway

Between Asterisk and Quintum if I use G729 RFC2833 DTMF works with no
issues, however if I use uLaw this is where there is a problem. For
some reason the Quintum gateway does not support uLaw + RFC2833.

Also does not matter if I use Asterisk 1.2 or a grandstream or the
proverbial SIP tin can; The scenario is always the same.
On Jan 28, 2008 7:03 PM, Alex Balashov <abalashov at evaristesys.com> wrote:
Quote:

I think your best bet is to do a packet capture and look for RTP packets
with an RTP Event payload ("rtpevent" display filter).


On Mon, 28 Jan 2008, Andrew Joakimsen wrote:

Quote:
How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no
messages related to DTMF... or if I just do a global SIP debug for
that matter.... I am using RFC DTMF but it's not being passed to the
PSTN and I need to debug this further. I've tried to increase the
verbosity and the debug ('set debug n') and that didn't help either. I
assume this is because even RFC2833 sends the DTMF as RTP which
wouldn't show up anyways.... but how to troubleshoot DTMF issues?

_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : +1-678-954-0670
Direct : +1-678-954-0671


_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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