But this is my questions why it is sending invites again in 6-10 when the call is already established. <br>-Jai<br><br><br><br><div class="gmail_quote">On Sat, Sep 5, 2009 at 3:22 AM, Olle E. Johansson <span dir="ltr"><<a href="mailto:oej@edvina.net">oej@edvina.net</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>
5 sep 2009 kl. 09.06 skrev Jai Rangi:<br>
<div class="im"><br>
> Thank you for your response,<br>
> But we do get response from client (Step 2,3,4), the call is good,<br>
> audio DTMF everything works, except CDR is wrong; always 30-60<br>
> seconds more for each call.<br>
</div>In step 6-10, there's no reply from the client, unless you missed<br>
something.<br>
Turn on SIP debug and you'll see that Asterisk will time out and give<br>
up about the call.<br>
<font color="#888888"><br>
/O<br>
</font><div><div></div><div class="h5">><br>
><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<br>
> > Progress<br>
> > 4 0.046546 192.168.4.23 -> 192.168.3.222 SIP/SDP Status: 200 OK,<br>
><br>
><br>
> On Fri, Sep 4, 2009 at 11:55 PM, Olle E. Johansson <<a href="mailto:oej@edvina.net">oej@edvina.net</a>><br>
> wrote:<br>
><br>
> 5 sep 2009 kl. 04.58 skrev Jai Rangi:<br>
><br>
> > Hello,<br>
> ><br>
> > I have a issue between asterisk and windows based VoIP system<br>
> > (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><br>
> > , 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<br>
> > Progress<br>
> > 4 0.046546 192.168.4.23 -> 192.168.3.222 SIP/SDP Status: 200 OK,<br>
> > 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" target="_blank">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:<br>
> INVITE <a href="http://sip:1978525648@192.168.4.23:5060" target="_blank">sip:1978525648@192.168.4.23:5060</a><br>
> > , with session description<br>
> > 7 0.266230 192.168.3.222 -> 192.168.4.23 RTP Payload type=ITU-T<br>
> > G.729, SSRC=905761218, Seq=56540, Time=0<br>
> > 8 1.046087 192.168.3.222 -> 192.168.4.23 SIP/SDP Request:<br>
> INVITE <a href="http://sip:1978525648@192.168.4.23:5060" target="_blank">sip:1978525648@192.168.4.23:5060</a><br>
> > , with session description<br>
> > 9 2.046091 192.168.3.222 -> 192.168.4.23 SIP/SDP Request:<br>
> INVITE <a href="http://sip:1978525648@192.168.4.23:5060" target="_blank">sip:1978525648@192.168.4.23:5060</a><br>
> > , with session description<br>
> > 10 4.046102 192.168.3.222 -> 192.168.4.23 SIP/SDP Request:<br>
> INVITE <a href="http://sip:1978525648@192.168.4.23:5060" target="_blank">sip:1978525648@192.168.4.23:5060</a><br>
> > , with session description<br>
> ><br>
> > I disconnected the call, Receive BYe from Vendor, Asterisk<br>
> > acknowledge Bye and does not send Bye to the client. Few more<br>
> > invites from Asterisk to the client machine.<br>
> ><br>
> > 11 8.046123 192.168.3.222 -> 192.168.4.23 SIP/SDP Request:<br>
> INVITE <a href="http://sip:1978525648@192.168.4.23:5060" target="_blank">sip:1978525648@192.168.4.23:5060</a><br>
> > , with session description<br>
> > 12 16.046179 192.168.3.222 -> 192.168.4.23 SIP/SDP Request:<br>
> INVITE <a href="http://sip:1978525648@192.168.4.23:5060" target="_blank">sip:1978525648@192.168.4.23:5060</a><br>
> > , 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:<br>
> INVITE <a href="http://sip:1978525648@192.168.4.23:5060" target="_blank">sip:1978525648@192.168.4.23:5060</a><br>
> > , 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<br>
> > Progress<br>
> > 18 32.092666 192.168.4.23 -> 192.168.3.222 SIP/SDP Status: 200 OK,<br>
> > with session description<br>
> > 19 32.593335 192.168.4.23 -> 192.168.3.222 SIP/SDP Status: 200 OK,<br>
> > with session description<br>
> > 20 33.607552 192.168.4.23 -> 192.168.3.222 SIP/SDP Status: 200 OK,<br>
> > with session description<br>
> ><br>
> > I am using canreinvite=yes, (Must use that to avoid media going<br>
> > 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<br>
> > invites.<br>
><br>
> There's no response from the client phone.<br>
> No 100 trying, no 180 ringing or 200 OK.<br>
> We have to retransmit a few times and then just give up.<br>
><br>
> Your client needs to wake up and start responding.<br>
><br>
> Since the client was not responding, there never was a call to the<br>
> client and no need to send a BYE.<br>
><br>
> /O<br>
><br>
<br>
<br>
<br>
<br>
<br>
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