[Asterisk-Users] DTMF Modes

[Dave Wolven]

Hi Eric,

asterisk -r
sip debug

You are looking for something like this:
Via: SIP/2.0/UDP 216.234.109.219:5060
From: "Dave Wolven"
<sip:dave1 at XXX.XXX.XXX.XXX>;tag=003094c25f4c4340108494f8-09c2eec9
To: <sip:92486504701 at XXX.XXX.XXX.XXX;user=phone>;tag=as70d41e8c
Call-ID: 003094c2-5f4c2cd3-5f53a536-1f966d65 at 216.234.109.219
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Contact: <sip:92486504701 at XXX.XXX.XXX.XXX>
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 1529 1529 IN IP4 XXX.XXX.XXX.XXX
s=session
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 23280 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

You are looking for rtpmap 101 telephone-event/8000

This looks like it would be using named telephone events (rtp-nte 101)
cisco standard.

Or at least that is my guess, you could probably look at RFC2833.

Just a quick guess though;-)

Thanks
Dave



On Mon, 2003-04-21 at 19:02, Eric Wieling wrote:
> How can I determine what DTMF mode is being used on a SIP call?  I'd
> also like to see the DTMF info in real time (via some kind of debug)?
> 
> I'm having problems with a device that's plugged into a Cisco ATA-186
> not recognizing DTMF.
> 
> --Eric
> -- 
> 
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