16 apr 2012 kl. 15:31 skrev Matthew Jordan: > It's not a bug - decrementing the CSeq header field value is directly in > violation of RFC 3261. From section 22.2: > > When a UAC resubmits a request with its credentials after receiving a > 401 (Unauthorized) or 407 (Proxy Authentication Required) response, > it MUST increment the CSeq header field value as it would normally > when sending an updated request.
This only applies to the same dialog. The question here is if it is the same dialog. If it is, then the server indeed has a bug. Check the Call-ID and the from tag of both requests. /Olle > ----- Original Message ----- >> From: "Benoit Panizzon" <[email protected]> >> To: [email protected] >> Sent: Monday, April 16, 2012 7:12:09 AM >> Subject: [asterisk-users] Invite + decreasing sequence number => 500 Error? >> >> Hi out there >> >> We have a strange Problem here with invites. >> >> We observe this SIP conversation. >> >> C3 PBX <-> Asterisk >> >> Case 1. Sequence Numer always increasing: >> >> => Invite >> <= 401 Unauthenticated >> => Invite+auth with sequence number > previous Invite. >> <= 100 Trying etc. Works OK. >> >> Case 2. Sequence Number decreasing. >> >> => Invite >> <= 401 Unauthenticated >> => Invite+auth with sequence number < previous Invite. >> <= 500 ERROR >> >> I was browsing the SIP rfc and I cannot find any clue if in this case >> the >> sequence numbers must be increasing (the C3 PBX is wrong) or if I >> have sumbled >> over an asterisk bug. >> >> Is there anyone who knows? >> >> Benoit Panizzon >> -- >> I m p r o W a r e A G - >> ______________________________________________________ >> >> Zurlindenstrasse 29 Tel +41 61 826 93 07 >> CH-4133 Pratteln Fax +41 61 826 93 02 >> Schweiz Web http://www.imp.ch >> ______________________________________________________ >> >> -- >> _____________________________________________________________________ >> -- 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 --- [email protected] - http://edvina.net The final Asterisk SIP Masterclass, June 11-15 in Barcelona, Spain. - Register today! -- _____________________________________________________________________ -- 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
