Hello, if this is the complete sip trace the UAS (client) have to reply with an ACK after the 401 response from asterisk. thats why asterisk thinks the request is still alive.
regards stefan Am 19.10.11 22:22, schrieb [email protected]: > Hallo, > > any idea what's wrong with that invite?? > help would be greatly appreciated! > > thanks > > Markus > > > U XX.199.123.185:5060 -> XX.189.169.66:5060 > INVITE sip:[email protected] SIP/2.0..Via: SIP/2.0/UDP > 192.168.178.26:5060;rport;branch=z9hG4bK98099..Max-Forwards: 70..To: > <sip:07111234567 > @XX.189.169.66>..From: > <sip:[email protected]>;tag=z9hG4bK84110414..Call-ID: > [email protected]: 1 INVITE..Contact: <sip:[email protected] > 6>..Expires: 3600..User-Agent: mjsip stack 1.6..Content-Length: > 154..Content-Type: application/sdp....v=0..o=sip:[email protected] 0 0 > IN IP4 192.168.17 > 8.26..s=Session SIP/SDP..c=IN IP4 192.168.178.26..t=0 0..m=audio 21000 > RTP/AVP 0..a=rtpmap:0 PCMU/8000.. > # > U XX.189.169.66:5060 -> XX.199.123.185:5060 > SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP > 192.168.178.26:5060;branch=z9hG4bK98099;received=XX.199.123.185;rport=5060..From: > <sip:[email protected]>;tag > =z9hG4bK84110414..To: > <sip:[email protected]>;tag=as76b40635..Call-ID: > [email protected]: 1 INVITE..User-Agent: Asterisk PBX 1 > .6.0.9..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, > NOTIFY..Supported: replaces, timer..WWW-Authenticate: Digest > algorithm=MD5, realm="a > sterisk", nonce="59645374"..Content-Length: 0.... > # > U XX.199.123.185:5060 -> XX.189.169.66:5060 > INVITE sip:[email protected] SIP/2.0..Via: SIP/2.0/UDP > 192.168.178.26:5060;rport;branch=z9hG4bK98099..Max-Forwards: 70..To: > <sip:07111234567 > @XX.189.169.66>..From: > <sip:[email protected]>;tag=z9hG4bK84110414..Call-ID: > [email protected]: 2 INVITE..Contact: <sip:[email protected] > 6>..Expires: 3600..User-Agent: mjsip stack 1.6..Authorization: Digest > username="12", realm="asterisk", nonce="59645374", > uri="sip:[email protected] > 69.66", algorithm=MD5, > response="b2b86ac54de0f1644da86bc5063e6a21"..Content-Length: > 154..Content-Type: application/sdp....v=0..o=sip:[email protected] 0 > 0 IN IP4 192.168.178.26..s=Session SIP/SDP..c=IN IP4 > 192.168.178.26..t=0 0..m=audio 21000 RTP/AVP 0..a=rtpmap:0 PCMU/8000.. > # > U XX.189.169.66:5060 -> XX.199.123.185:5060 > SIP/2.0 491 Request Pending..Via: SIP/2.0/UDP > 192.168.178.26:5060;branch=z9hG4bK98099;received=XX.199.123.185;rport=5060..From: > <sip:[email protected]>; > tag=z9hG4bK84110414..To: > <sip:[email protected]>;tag=as76b40635..Call-ID: > [email protected]: 2 INVITE..User-Agent: Asterisk PB > X 1.6.0.9..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, > NOTIFY..Supported: replaces, timer..Content-Length: 0.... > # > > > -- > _____________________________________________________________________ > -- 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 -- Für weitere Fragen stehen wir gerne unter [email protected] oder 059944 - 2440 zur Verfügung. Mit freundlichen Grüssen -- Stefan Schmidt Teamleiter VOIP // [email protected] // Tel 059944-2440// ------------------------------------------------- SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // ------------------------------------------------- -- _____________________________________________________________________ -- 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
