Hi List I have come over a codec negotiation issue.
A (asterisk) is sending in INVITE containing * opus (type 107) * g722 * alaw (type 8) B answers with 183 containing SDP * alaw a=sendrecv B then answer the call with 200 and NO SDP I suppose that result in B telling us, it only support alaw. But 'set rtp debug on' show B sending type 8 and A sending type 107. As the remote only announced to be capable of 8, shouldn't asterisk send type 8? Or even send a Re-Invite to tell it switches to alaw? Also reading: https://wiki.asterisk.org/wiki/display/AST/PJSIP+Advanced+Codec+Negotiation does not explain what I see. -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________ -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
