Hello! Thank you for that piece of information. Which RFC does it state that the audio name is "G729"?
Thanks, Elliot On Thu, Jul 2, 2009 at 12:16 AM, Kevin P. Fleming<[email protected]> wrote: > Elliot Murdock wrote: >> Hello! >> >> I have a sip device that is sending in the SDP: >> >> rtpmap:98 g729a >> >> It does not seem like Asterisk is negotiating the codec properly, >> because while the call rings, the rtp lines fail. However, on other >> sip devices that have "rtpmap:18 g729" in their SDP, things work fine >> with Digium's commercial g729 license. >> >> How do I get "98 g729a" recognized by Asterisk? > > You don't. That's not a standards-compliant way of reporting G.729A in > SDP. The RFC says it should be 'G729', but Asterisk also accepts 'G.729' > and 'G729A'. It does not accept any lowercase form of the codec name. > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > skype: kpfleming | jabber: [email protected] > Check us out at www.digium.com & www.asterisk.org > > _______________________________________________ > -- 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 > _______________________________________________ -- 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
