[asterisk-users] Urgent: DTMF does not work with rtpmap:101 telephone-event/8000

[Olle E Johansson]

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