But this is my questions why it is sending invites again in 6-10 when the call is already established. -Jai
On Sat, Sep 5, 2009 at 3:22 AM, Olle E. Johansson <o...@edvina.net> wrote: > > 5 sep 2009 kl. 09.06 skrev Jai Rangi: > > > Thank you for your response, > > But we do get response from client (Step 2,3,4), the call is good, > > audio DTMF everything works, except CDR is wrong; always 30-60 > > seconds more for each call. > In step 6-10, there's no reply from the client, unless you missed > something. > Turn on SIP debug and you'll see that Asterisk will time out and give > up about the call. > > /O > > > > > > 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, > > > > > > On Fri, Sep 4, 2009 at 11:55 PM, Olle E. Johansson <o...@edvina.net> > > wrote: > > > > 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:1978525...@192.168.4.23 <sip%3a1978525...@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:1978525...@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:1978525...@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:1978525...@192.168.4.23:5060 > > > , with session description > > > 9 2.046091 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: > > INVITE sip:1978525...@192.168.4.23:5060 > > > , with session description > > > 10 4.046102 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: > > INVITE sip:1978525...@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:1978525...@192.168.4.23:5060 > > > , with session description > > > 12 16.046179 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: > > INVITE sip:1978525...@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:6056929...@192.168.3.222 <sip%3a6056929...@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:1978525...@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 > > > > > > > > _______________________________________________ > -- 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 >
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