RE: [Asterisk-Users] Passing DTMF

[Brian Rathman]

Title: Re: [Asterisk-Users] Passing DTMF

I have tried every combination of codec and dtmfmode. I can hear the dtmf tone on the far end phone, it just appears to be to short. Is there a way to increase the duration of the tone?

-----Original Message-----
From: Eric Wieling [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 06, 2004 3:10 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Passing DTMF

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/

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