Hi This rtpmap is in the dynamic range and should be determined by the by the UAC sending the INVITE message. Thus SFLphone should automatically select 101 for the DTMF rtp payload type. What version of SFLphone are you using since I think this issue is supposed to be fixed in the latest version.
Regards, Alexandre ----- Original Message ----- From: "John A. Sullivan III" <[email protected]> To: "Simon MORIN" <[email protected]> Cc: [email protected] Sent: Wednesday, December 21, 2011 10:17:03 AM Subject: Re: [SFLphone] [SFLPhone] ] Problem with DTMF payload code On Wed, 2011-12-21 at 09:22 +0100, Simon MORIN wrote: > Dear all, > > > > I have a problem using SFLPhone with an Alcatel OXE v10 IPBX. When I > receive a call on SFLPhone from an other telephone, the communication > hang up automaticaly when I accept the call in SFLPhone. The other > telephone get the communication cut, but not SFLPhone. > > > > I did a frame capture (eventually, I can send the Wireshark file). The > exchanged frames are : > > > > From IPBX : INVITE with SDP > > From SFL : Status 180 Ringing > > From SFL : Status OK with SDP > > From SFL : Status OK with SDP > > From SFL : Status OK with SDP > > From SFL : Status OK with SDP > > From SFL : Request BYE > > From IPBX : Status 404 > > > > When going deeper in the INVITE and in the Status OK, I saw this : > > > > - In the incoming INVITE from the IPBX, the SDP contains : > > v=0 > > o=OXE 123456 123456 IN IP4 192.168.20.36 > > s=abs > > c=IN IP4 192.168.20.41 > > t=0 0 > > m=audio 32048 RTP/AVP 8 97 > > a=sendrecv > > a=rtpmap:8 PCMA/8000 > > a=ptime:20 > > a=maxptime:30 > > a=rtpmap:97 telephone-event/8000 > > > > - In the OK response from the SFLPhone, the SDP contains : > > v=0 > > o=sflphone 123456 1 IN IP4 192.168.40.43 > > s=sflphone > > c=IN IP4 192.168.40.43 > > t=0 0 > > m=audio 26562 RTP/AVP 8 > > a=rtpmap:8 PCMA/8000 > > a=sendrecv > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > > > > > It does send the "200 OK" several time without getting any answer. I > compared with an other softphone, and the differences I saw were that > this other softphone send the telephone-event as 97 and not as 101. > > Is there a way to use the code "97" in SFLPhone for the > telephone-event instead of 101 ? > > > > Also, is it possible in the "m=" line to add the telephone-event as a > secondary audio codec ? <snip> Interesting. I wonder if that is related to our problem where our Asterisk server does not understand any of the DTMF tones sent by SFLPhone. We are thus completely unable to use it - John _______________________________________________ SFLphone mailing list [email protected] http://lists.savoirfairelinux.net/mailman/listinfo/sflphone _______________________________________________ SFLphone mailing list [email protected] http://lists.savoirfairelinux.net/mailman/listinfo/sflphone
