5 sep 2009 kl. 04.58 skrev Jai Rangi: > Hello, > > I have a issue between asterisk and windows based VoIP system > (Client). > > Vendor SIP Server --> My asterisk --> Client > Here is ethereal trace between asterisk and client. > > 1 0.000000 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE > sip:[email protected] > , with session description > 2 0.042380 192.168.4.23 -> 192.168.3.222 SIP Status: 100 Trying > 3 0.044235 192.168.4.23 -> 192.168.3.222 SIP Status: 183 Session > Progress > 4 0.046546 192.168.4.23 -> 192.168.3.222 SIP/SDP Status: 200 OK, > with session description > 5 0.046752 192.168.3.222 -> 192.168.4.23 SIP Request: ACK > sip:[email protected]:5060 > So far so good, call is established and audio conversations starts. > > But next my asterisk is sending Invite again and again and again, > > 6 0.047036 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE > sip:[email protected]:5060 > , with session description > 7 0.266230 192.168.3.222 -> 192.168.4.23 RTP Payload type=ITU-T > G.729, SSRC=905761218, Seq=56540, Time=0 > 8 1.046087 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE > sip:[email protected]:5060 > , with session description > 9 2.046091 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE > sip:[email protected]:5060 > , with session description > 10 4.046102 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE > sip:[email protected]:5060 > , with session description > > I disconnected the call, Receive BYe from Vendor, Asterisk > acknowledge Bye and does not send Bye to the client. Few more > invites from Asterisk to the client machine. > > 11 8.046123 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE > sip:[email protected]:5060 > , with session description > 12 16.046179 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE > sip:[email protected]:5060 > , with session description > > After a 30 second wait, asterisk receive Bye from Client. > > 13 24.253811 192.168.4.23 -> 192.168.3.222 SIP Request: BYE > sip:[email protected] > 14 24.253975 192.168.3.222 -> 192.168.4.23 SIP Status: 200 OK > 15 32.046319 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE > sip:[email protected]:5060 > , with session description > 16 32.085897 192.168.4.23 -> 192.168.3.222 SIP Status: 100 Trying > 17 32.090654 192.168.4.23 -> 192.168.3.222 SIP Status: 183 Session > Progress > 18 32.092666 192.168.4.23 -> 192.168.3.222 SIP/SDP Status: 200 OK, > with session description > 19 32.593335 192.168.4.23 -> 192.168.3.222 SIP/SDP Status: 200 OK, > with session description > 20 33.607552 192.168.4.23 -> 192.168.3.222 SIP/SDP Status: 200 OK, > with session description > > I am using canreinvite=yes, (Must use that to avoid media going > through my asterisk server. > I dont have any issue if asterisk send call to another asterisk box. > > Can some one please shed some light why asterisk is sending multiple > invites.
There's no response from the client phone. No 100 trying, no 180 ringing or 200 OK. We have to retransmit a few times and then just give up. Your client needs to wake up and start responding. Since the client was not responding, there never was a call to the client and no need to send a BYE. /O --- [email protected] - http://edvina.net Open Unified Communication - building platforms with SIP and XMPP From PBX to large scale implementations for carriers. Contact us today! _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
