On Wed, 2010-01-13 at 10:12 -0500, Kristian Kielhofner wrote: > On Wed, Jan 13, 2010 at 2:43 AM, [email protected] > <[email protected]> wrote: > > Thanks for that. Looking at the RTP packets I can see two types as you > > point out. The first appears to be the audio: > > > > Real-Time Transport Protocol > > 10.. .... = Version: RFC 1889 Version (2) > > Payload type: ITU-T G.711 PCMU (0) > > > > And as you say, the DTMF events are clear to see: > > RFC 2833 RTP Event > > Event ID: DTMF One 1 (1) > > ..00 1010 = Volume: 10 > > > > So, as these can be seen in the stream, do I need to tell Asterisk to > > detect these? Does it not do that when I set: dtmfmode=rfc2833 > > ??? > > There are some pretty widely recognized RFC2833 compatibility issues > in the SIP/RTP world. I had a nasty feeling something like that was coming :-(
> Which version of Asterisk are you using? Asterisk 1.6.1.11 > Do > you know what kind of equipment your carrier is using? If they are > using Asterisk you can try to add rfc2833compensate=yes to their peer > entry in sip.conf. I've not been able to get that out of them, but I don't *think* It's Asterisk based because they say: "Unfortunately, our assistance with Asterisk is extremely limited. For configuration problems you will have to rely on other sources." [http://www.sipgate.co.uk/faq/index.php?do=displayArticle&article=540&qw=asterisk] > > >> > >> The SIP debug, however, will tell you if the remote end is configured > >> to use RFC2833 or not. That's why I was telling you to look for > >> telephone-event in the INVITE from your provider. Keep in mind SIP > >> (most likely) runs over UDP between you and your provider, not TCP. > >> > > I see a 'telephone-event' : > > > > a=rtpmap:101 telephone-event/8000 > > > > That's all you need to know. They are configured for RFC2833 and > they're sending RFC2833. I appreciate this is a 'how long is a piece of string question Kristian, but is there likely to be a way I can fix this? -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
