25 maj 2007 kl. 06.40 skrev JK:
Hello asterisk-users list.
I have been scratching my head for almost a week. We are trying to
set a service with a company (ip=XXX.XXX.XXX.XXX) and dtmf is not
working.
In our scenario the SP is sending call to our ser server and ser
is forwarding the call to asterisk. In the asterisk debug I can see
the DTMF keys are coming but ivr does not recognice those keys at
all. I can see this in the debug. We are using ulaw and alaw for
codec.
May 24 20:14:00 DEBUG[26803] rtp.c: Sending dtmf: 49 (1), at
XXX.XXX.XXX.XXX
May 24 20:14:01 DEBUG[26803] rtp.c: Sending dtmf: 49 (1), at
XXX.XXX.XXX.XXX
May 24 20:14:01 DEBUG[26803] rtp.c: Sending dtmf: 49 (1), at
XXX.XXX.XXX.XXX
Voice part works great. I mean if I forward that call to asterisk
sip user we can talk.
Every thing is working great with other SP. The only difference I
can see is the rtpmap:101 telephone-event/8000.
With the working SP the rtpmap is rtpmap:100 telephone-event/8000.
Your debug did not have any SIP messages. I need to see the INVITE
and the 200 OK. Thanks.
/O
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