Jai Rangi
2009-Sep-05 02:58 UTC
[asterisk-users] Need some help/Suggestions for multiple invites from Asterisk
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:1978525648 at 192.168.4.23 <sip%3A1978525648 at 192.168.4.23>, 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:1978525648 at 192.168.4.23:5060 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:1978525648 at 192.168.4.23:5060, 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:1978525648 at 192.168.4.23:5060, with session description 9 2.046091 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE sip:1978525648 at 192.168.4.23:5060, with session description 10 4.046102 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE sip:1978525648 at 192.168.4.23:5060, 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:1978525648 at 192.168.4.23:5060, with session description 12 16.046179 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE sip:1978525648 at 192.168.4.23:5060, 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:6056929587 at 192.168.3.222 <sip%3A6056929587 at 192.168.3.222> 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:1978525648 at 192.168.4.23:5060, 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. Best, -Jai -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090904/4c000d78/attachment.htm
Olle E. Johansson
2009-Sep-05 06:55 UTC
[asterisk-users] Need some help/Suggestions for multiple invites from Asterisk
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:1978525648 at 192.168.4.23 > , 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:1978525648 at 192.168.4.23:5060 > 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:1978525648 at 192.168.4.23:5060 > , 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:1978525648 at 192.168.4.23:5060 > , with session description > 9 2.046091 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE sip:1978525648 at 192.168.4.23:5060 > , with session description > 10 4.046102 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE sip:1978525648 at 192.168.4.23:5060 > , 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:1978525648 at 192.168.4.23:5060 > , with session description > 12 16.046179 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE sip:1978525648 at 192.168.4.23:5060 > , 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:6056929587 at 192.168.3.222 > 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:1978525648 at 192.168.4.23:5060 > , 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 --- oej at edvina.net - http://edvina.net Open Unified Communication - building platforms with SIP and XMPP From PBX to large scale implementations for carriers. Contact us today!