I had a similar problem that seems to be caused by the DTMF tone lengths
being to short. Try this:
Asterisk generates DTMF tones in do_senddigit() in the file channel.c.
The tones are defined in a const char array called dtmf_tones[]. Each
DTMF tone is a string that looks something like:
Hi Mr. Evil,
I'm not sure if the problem that I am describing relates to the problem
that you are having. It seems that when you press a key on a SIP phone
that is set for inband DTMF, asterisk absorbs the tones until you
release the key. This way if you are using DTMF to do things like
Asterisk generates DTMF tones in do_senddigit() in the file channel.c.
The tones are defined in a const char array called dtmf_tones[]. Each
DTMF tone is a string that looks something like:
!941+1336/100,!0/100, /* 0 */
The part that reads !941+1336/100 is the part that you want. Change
I am trying to place an analog outbound call from a Sipura SPA-841
through a * server with a TDM400P and 4 FXO's. When I call in from an
analog line everything works fine, I can talk over the SIP phone. When
I call out, * says:
== Spawn extension (from-sip, [phonenumber], 1) exited non-zero
__do_dtmf() with all
of the digits that I would like it to dial, but there is no sound on the
wire. Any ideas?
Thanks,
Rob
Rob Tarte wrote:
I am trying to place an analog outbound call from a Sipura SPA-841
through a * server with a TDM400P and 4 FXO's. When I call in from an
analog line everything
Brown wrote:
Try removing the g from the dial command:
exten = _X.,1,Dial(Zap/1/${EXTEN},60)
exten = _X.,2,Hangup ;
exten = _NXXX,1,Dial(Zap/1)
Simon Brown
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Rob Tarte
Sent: Wednesday, 2 February 2005 16
