I've encountered a similar problem with Cisco equipment. The Cisco proxy was not replying to Asterisk with an ACK after * sent an OK.<br><br>Since version 1.2.14, * was changed so that not receiving an ACK to an OK is considered a FATAL error.
<br><br>The specific change that causes this problem is in sip_answer() in chan_sip.c:<br><br>res = transmit_response_with_sdp(p, "200 OK", &p->initreq, 2);<br><br>Changing the 2 to a 1 will probably fix it. Note that this is NOT a bug in * but improper implementations--either caused by latency, or a software bug (not sending an ACK). Perhaps it might be beneficial to have an option in
sip.conf to change how * handles not receiving an ACK? I know... it's someone else's problem, but might help those of us stuck with buggy implementations in production environments. :)<br><br>Brian.<br><br><br><div>
<span class="gmail_quote">On 4/12/07, <b class="gmail_sendername">Joao Pereira</b> <<a href="mailto:joao.pereira@fccn.pt">joao.pereira@fccn.pt</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hello<br>Thanks a lot for your reply.<br>Im now using asterisk-1.2.10 and the problem disappeared.<br>Thanks<br>regards<br>Joao Pereira<br><br><br>Edoardo Serra wrote:<br>> Same to me !!<br>><br>> Calls from OpenSER to Asterisk
<br>><br>> It happens only with Asterisk versions >= 1.2.14<br>><br>> I'm going to capture some traffic<br>><br>> Tnx for help<br>><br>> Regards<br>><br>> Alex Balashov ha scritto:<br>
>><br>>> Joao,<br>>><br>>> It sounds like the proxy is not acknowledging the Asterisk's<br>>> processing of the INVITE, but I could be wrong. It would be helpful<br>>> to supply a packet capture between OpenSER and Asterisk so we could
<br>>> see the setup flow.<br>>><br>>> Thanks,<br>>><br>>> -- Alex<br>>><br>>> On Tue, 10 Apr 2007, Joao Pereira said something to this effect:<br>>><br>>>> Hello<br>
>>> My asterisk is receiving calls from OpenSER but all calls hangup in<br>>>> 20 seconds.<br>>>> This only happens because Im using Asterisk2Billing's AGI (without<br>>>> A2Billing it doesnt hang up).
<br>>>> does someone knows whats the problem??<br>>>><br>>>> Here is my Asterisk debug:<br>>>> (xxx.xxx.xxx.xxx -> the phone's IP)<br>>>><br>>>><br>>>>
<br>>>> Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread:<br>>>> Unable to spawn mp3player<br>>>> Apr 10 02:04:18 NOTICE[8349]: rtp.c:331 process_rfc3389: Comfort<br>>>> noise support incomplete in Asterisk (RFC 3389). Please turn off on
<br>>>> client if possible. Client IP: xxx.xxx.xxx.xxx<br>>>> Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1228 retrans_pkt: Maximum<br>>>> retries exceeded on transmission<br>>>> <a href="mailto:CAB5E4C9-8B95-42D4-9A93-FB996EDC43A2@xxx.xxx.xxx.xxx">
CAB5E4C9-8B95-42D4-9A93-FB996EDC43A2@xxx.xxx.xxx.xxx</a> for seqno 12282<br>>>> (Critical Response)<br>>>> Apr 10 02:04:19 WARNING[7007]: chan_sip.c:1245 retrans_pkt: Hanging<br>>>> up call <a href="mailto:CAB5E4C9-8B95-42D4-9A93-FB996EDC43A2@xxx.xxx.xxx.xxx">
CAB5E4C9-8B95-42D4-9A93-FB996EDC43A2@xxx.xxx.xxx.xxx</a> - no<br>>>> reply to our critical packet.<br>>>> Apr 10 02:06:01 NOTICE[8360]: rtp.c:331 process_rfc3389: Comfort<br>>>> noise support incomplete in Asterisk (RFC 3389). Please turn off on
<br>>>> client if possible. Client IP: xxx.xxx.xxx.xxx<br>>>><br>>>><br>>>> Thanks for the help<br>>>> Regards<br>>>> Joao Pereira<br>>>><br>>>><br>>>> _______________________________________________
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<br>>>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>>>><br>>><br>>> --<br>>> Alex Balashov <<a href="mailto:sasha@presidium.org">
sasha@presidium.org</a>><br>>> _______________________________________________<br>>> --Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br>>><br>>> asterisk-users mailing list
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