Hello,<br><br>I have a issue between asterisk and windows based VoIP system (Client). <br><br>Vendor SIP Server --> My asterisk --> Client <br>Here is ethereal trace between asterisk and client. <br><br>1 0.000000 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE <a href="mailto:sip%3A1978525648@192.168.4.23">sip:1978525648@192.168.4.23</a>, with session description<br>
2 0.042380 192.168.4.23 -> 192.168.3.222 SIP Status: 100 Trying<br> 3 0.044235 192.168.4.23 -> 192.168.3.222 SIP Status: 183 Session Progress<br> 4 0.046546 192.168.4.23 -> 192.168.3.222 SIP/SDP Status: 200 OK, with session description<br>
5 0.046752 192.168.3.222 -> 192.168.4.23 SIP Request: ACK <a href="http://sip:1978525648@192.168.4.23:5060">sip:1978525648@192.168.4.23:5060</a><br>So far so good, call is established and audio conversations starts. <br>
<br>But next my asterisk is sending Invite again and again and again, <br><br> 6 0.047036 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE <a href="http://sip:1978525648@192.168.4.23:5060">sip:1978525648@192.168.4.23:5060</a>, with session description<br>
7 0.266230 192.168.3.222 -> 192.168.4.23 RTP Payload type=ITU-T G.729, SSRC=905761218, Seq=56540, Time=0<br> 8 1.046087 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE <a href="http://sip:1978525648@192.168.4.23:5060">sip:1978525648@192.168.4.23:5060</a>, with session description<br>
9 2.046091 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE <a href="http://sip:1978525648@192.168.4.23:5060">sip:1978525648@192.168.4.23:5060</a>, with session description<br> 10 4.046102 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE <a href="http://sip:1978525648@192.168.4.23:5060">sip:1978525648@192.168.4.23:5060</a>, with session description<br>
<br>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. <br><br> 11 8.046123 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE <a href="http://sip:1978525648@192.168.4.23:5060">sip:1978525648@192.168.4.23:5060</a>, with session description<br>
12 16.046179 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE <a href="http://sip:1978525648@192.168.4.23:5060">sip:1978525648@192.168.4.23:5060</a>, with session description<br><br>After a 30 second wait, asterisk receive Bye from Client. <br>
<br> 13 24.253811 192.168.4.23 -> 192.168.3.222 SIP Request: BYE <a href="mailto:sip%3A6056929587@192.168.3.222">sip:6056929587@192.168.3.222</a><br> 14 24.253975 192.168.3.222 -> 192.168.4.23 SIP Status: 200 OK<br>
15 32.046319 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE <a href="http://sip:1978525648@192.168.4.23:5060">sip:1978525648@192.168.4.23:5060</a>, with session description<br> 16 32.085897 192.168.4.23 -> 192.168.3.222 SIP Status: 100 Trying<br>
17 32.090654 192.168.4.23 -> 192.168.3.222 SIP Status: 183 Session Progress<br> 18 32.092666 192.168.4.23 -> 192.168.3.222 SIP/SDP Status: 200 OK, with session description<br> 19 32.593335 192.168.4.23 -> 192.168.3.222 SIP/SDP Status: 200 OK, with session description<br>
20 33.607552 192.168.4.23 -> 192.168.3.222 SIP/SDP Status: 200 OK, with session description<br><br>I am using canreinvite=yes, (Must use that to avoid media going through my asterisk server. <br>I dont have any issue if asterisk send call to another asterisk box. <br>
<br>Can some one please shed some light why asterisk is sending multiple invites. <br><br>Best,<br>-Jai<br><br>