Does anyone know how I can pass dtmf digits from a SNOM 200 to a cisco AS5300 with * in the media stream. Unfortunately, the only way I can get the calls to connect is with t or T at the end of the Dial() statement and then that picks off the dtmf digits. I have tried the canreinvite=yes on both the phone peer and the gateway peer and I still have to add the T to the Dial statement to make the call complete. Any suggestions??? Thanks, Brian
[Asterisk-Users] Passing DTMFJust to follow up, it does not matter what
codec I use, and when I listen to the call on the far end, I can hear a very
quick blip that sounds like the correct tone, but it is not long enough for
an IVR to recognize. Is there a way to boost the length of this tone in
Asterisk? Any help would be greatly appreciated.
-----Original Message-----
From: Brian J. Rathman [mailto:brian@ilk.com]
Sent: Tuesday, April 06, 2004 1:29 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Passing DTMF
Does anyone know how I can pass dtmf digits from a SNOM 200 to a cisco
AS5300 with * in the media stream. Unfortunately, the only way I can get
the
calls to connect is with t or T at the end of the Dial() statement and
then
that picks off the dtmf digits. I have tried the canreinvite=yes on both
the
phone peer and the gateway peer and I still have to add the T to the Dial
statement to make the call complete. Any suggestions???
Thanks,
Brian
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On Tue, 2004-04-06 at 12:29, Brian Rathman wrote:> Does anyone know how I can pass dtmf digits from a SNOM 200 to a cisco > AS5300 with * in the media stream. Unfortunately, the only way I can get the > calls to connect is with t or T at the end of the Dial() statement and then > that picks off the dtmf digits. I have tried the canreinvite=yes on both the > phone peer and the gateway peer and I still have to add the T to the Dial > statement to make the call complete. Any suggestions???cantrinvite=yes tells asterisk to, if it can, remove itself from the media stream. T and t and r and many other Dial options tells Asterisk to stay in the media stream so it can listen to the DTMF. None of this has ANYTHING to do with passing DTMF between the two endpoints (except of course passing # for t or T). If you cannot pass DTMF between the two endpoints then something ELSE is wrong. Maybe you are trying to use inband DTMF with a compressed codec. Inband DTMF will only work with ulaw or alaw codecs. --Eric -- Useful Asterisk Docs (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unofficial Links") and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/
Eric Wieling wrote:> On Tue, 2004-04-06 at 12:29, Brian Rathman wrote: > >>Does anyone know how I can pass dtmf digits from a SNOM 200 to a cisco >>AS5300 with * in the media stream. Unfortunately, the only way I can get the >>calls to connect is with t or T at the end of the Dial() statement and then >>that picks off the dtmf digits. I have tried the canreinvite=yes on both the >>phone peer and the gateway peer and I still have to add the T to the Dial >>statement to make the call complete. Any suggestions??? > > > cantrinvite=yes tells asterisk to, if it can, remove itself from the > media stream. T and t and r and many other Dial options tells Asterisk > to stay in the media stream so it can listen to the DTMF. None of this > has ANYTHING to do with passing DTMF between the two endpoints (except > of course passing # for t or T). If you cannot pass DTMF between the > two endpoints then something ELSE is wrong. Maybe you are trying to use > inband DTMF with a compressed codec. Inband DTMF will only work with > ulaw or alaw codecs....or the problem is, as hinted, that Asterisk sends a short dtmf. Regardless of what it receives into the sip channel, Asterisk sends a 250 ms DTMF signal out (if my memory is correct). You can check in chan_sip.c The dtmf setting sets what Asterisk sends to that peer/user. /O