<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:times new roman, new york, times, serif;font-size:12pt"><div>Hope someone can help.<br><br>I have a situation where asterisk is sending a SIP CANCEL message before the Dial() timeout has hit. It doesn't always do it.<br><br>Normally, we send an INVITE to the ITSP. They respond with a 100 Trying, then a 180 Ringing, or 183 Session Progress. It seems to be at this point that Asterisk starts the dial timer. Normally, when no more replies have been received by the dial timeout, Asterisk sends a CANCEL message. That's all fine, and when this happens, this is what appears on the console:<br><br> -- Called 919431555555@teleglobe<br> -- SIP/teleglobe-09879188 is making progress passing it to SIP/teleglobe-09876568<br> -- Nobody picked up in 40000 ms<br> -- Executing
PlayTones("SIP/teleglobe-09876568", "congestion") in new stack<br><br>However, when asterisk sends the CANCEL earlier then this, this is what appears on the console:<br><br> -- SIP/teleglobe-09879188 is making progress passing it to SIP/teleglobe-09876568<br> == Spawn extension (default, callback, 7) exited non-zero on 'SIP/teleglobe-09876568'<br>Jan 9 01:16:34 WARNING[5719]: channel.c:781 channel_find_locked: Avoided initial deadlock for '0x97f24d8', 10 retries!<br><br>Does anyone know what the deadlock message is all about? It is ocurring quite frequently.<br>This is Asterisk 1.2.14.<br><br>Thanks,<br>Doug<br><br><br><br></div></div><br>
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