Hi Kevin, will grab a sip trace today and post it up. -- Thanks, Phil
----- Original Message ----- > On 01/23/2012 09:48 AM, --[ UxBoD ]-- wrote: > > Hi, > > > > I am attempting to make a SIP call between an Asterisk 10 server > > and an > > Asterisk 1.8 system but when it goes to VM and the first prompt > > plays > > the line drops and I see on the V10 console: > > > > [Jan 23 15:47:04] WARNING[7859]: chan_sip.c:8944 process_sdp: > > Insufficient information for SDP (m= not found) > > > > Any ideas please as the codecs at each end look okay ? > > There is either an error in the SDP generated by one end, or an error > in > the parser at the end receiving it. Posting the actual messages > involved > would help tremendously, because otherwise 'ideas' would be just > guesses. > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > Jabber: [email protected] | SIP: [email protected] | Skype: > kpfleming > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at www.digium.com & www.asterisk.org > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
