I can confirm the same error message... i haven't done nearly the amount of debuggin you have but it's the exact same error message i receive when i use a software based SIP phone connecting to another internal software SIP phone... some times it's twinkle to xlite some times xlite to xlite and some times twinkle to twinkle ...
<br><br>That's about all that i can confirm :) hope you get some help cuz i was also looking for some info on what the issue was.<br><br><div><span class="gmail_quote">On 5/8/07, <b class="gmail_sendername"><a href="mailto:chris@cgb1911.mine.nu">
chris@cgb1911.mine.nu</a></b> <<a href="mailto:chris@cgb1911.mine.nu">chris@cgb1911.mine.nu</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;">
(repost - can anyone confirm whether they've seen this before, or have<br>any tipes in debugging it?)<br><br>Hi Everyone,<br><br>I was hoping someone might know why I am experiencing a problem with<br>Asterisk logging the event:
<br><br>[May 3 12:07:41] WARNING[30371] chan_sip.c: Maximum retries exceeded on transmission <a href="mailto:03f007af2b15cd0b54b0c368265d97be@sip.externalprovider.com">03f007af2b15cd0b54b0c368265d97be@sip.externalprovider.com
</a> for seqno 669371069 (Critical Response)<br><br>This is happening after:<br> - call is setup, 2 way audio<br> - call can function correctly for up to 5 minutes, with the external<br> provider re-inviting every 1 minute
<br><br>When the problem happens<br> - external peer re-invites asterisk<br> - asterisk sends 200 OK<br> - external peer sends ACK<br> - asterisk retransmits 200 OK<br> - external peer sends ack<br> - ..<br> - asterisk retransmits 200 OK (Retransmitting #6)
<br> - external peer sends ack<br> - Asterisk logs the above message about maximum retries exceeded,<br> and sends BYE to the inside SIP UA.<br><br><br>The network configuration is as follows:<br> phone <--> alternative SIP server <--> Asterisk <-NAT-> External peer
<br><br>The alternative SIP server is not a B2BUA, just SIP proxy. Now,<br>sometimes a call can work without any problems, but not as often as<br>when the above symptoms are experienced.<br><br>The references I've found online about this type of problem suggest
<br>NAT as being the culprit, but in this case, Asterisk is logging it's<br>reception of the ACK but deciding to ignore it and retransmit the<br>200 OK anyhow. I'm guessing in other cases people suspect is' NAT
<br>because they believe SIP isn't getting back trhough after a period of<br>time.<br><br>I was using 1.4.2, but found this changelog today for 1.4.3:<br><br><a href="ftp://ftp.digium.com/pub/asterisk/releases/ChangeLog-1.4.3">
ftp://ftp.digium.com/pub/asterisk/releases/ChangeLog-1.4.3</a><br><br>2006-09-30 16:12 +0000 [r44068-44078] Paul Cadach <<a href="mailto:paul@odt.east.telecom.kz">paul@odt.east.telecom.kz</a>><br> * channels/chan_sip.c: Found some buggy SIP clients (phones Planet
<br> VIP-153T firmware 1.0, Linksys PAP2 firmware 3.1.9(LSc)) which<br> sends ACK not on OK message only (when remote party answers) but<br> on RINGING message too, so when we send 200 OK message, we get<br> unidentified ACK message (because INVITE acknowledged on RINGING
<br> message already), so 200 OK retransmits within its retransmission<br> interval then call gets dropped. If someone else knows how to<br> provide workaround for such cases, please, fix it in correct way.<br> Thanks to ssh from #asteriskru for provide access to his box to
<br> study and fix this case.<br><br>I've upgraded to 1.4.4 but the problem still persists. The above<br>changelog doesn't sound exactly like what I"m experiencing but maybe<br>it's related.<br><br>Attached is my
sip.conf, extensions.conf, and (debug = 10) logs for<br>one example. I don't know what else might be needed to help anyone<br>assist me in this problem - let me know if I missed something.<br><br>It *feels* like an Asterisk bug but maybe a SIP expert can spot the
<br>problem in signalling/RFC conformance..<br><br>Thanks in advance,<br><br>Chris Bennett<br><br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com
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<br><br></blockquote></div><br>