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[dougmig33 at yahoo.com]

Hope someone can help.

I have a situation where asterisk is sending a SIP CANCEL message before the Dial() timeout has hit. It doesn't always do it.

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:

-- Called 919431555555 at teleglobe
-- SIP/teleglobe-09879188 is making progress passing it to SIP/teleglobe-09876568
-- Nobody picked up in 40000 ms
-- Executing PlayTones("SIP/teleglobe-09876568", "congestion") in new stack

However, when asterisk sends the CANCEL earlier then this, this is what appears on the console:

-- SIP/teleglobe-09879188 is making progress passing it to SIP/teleglobe-09876568
== Spawn extension (default, callback, 7) exited non-zero on 'SIP/teleglobe-09876568'
Jan 9 01:16:34 WARNING[5719]: channel.c:781 channel_find_locked: Avoided initial deadlock for '0x97f24d8', 10 retries!

Does anyone know what the deadlock message is all about? It is ocurring quite frequently.
This is Asterisk 1.2.14.

Thanks,
Doug





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[dougmig33 at yahoo.com]

Replying to myself.
I just noticed the deadlock message still displayed on the console at the end of a normal call, so the the deadlock message is not related to the early CANCEL

----- Original Message ----
From: Douglas Garstang <dougmig33 at yahoo.com>
To: asterisk-users at lists.digium.com
Sent: Tuesday, January 8, 2008 5:31:12 PM
Subject: [asterisk-users] Help! channel_find_deadlocked: Avoided initial deadlock for ...
Hope someone can help.

I have a situation where asterisk is sending a SIP CANCEL message before the Dial() timeout has hit. It doesn't always do it.

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:

-- Called 919431555555 at teleglobe
-- SIP/teleglobe-09879188 is making progress passing it to SIP/teleglobe-09876568
-- Nobody picked up in 40000 ms
-- Executing
PlayTones("SIP/teleglobe-09876568", "congestion") in new stack

However, when asterisk sends the CANCEL earlier then this, this is what appears on the console:

-- SIP/teleglobe-09879188 is making progress passing it to SIP/teleglobe-09876568
== Spawn extension (default, callback, 7) exited non-zero on 'SIP/teleglobe-09876568'
Jan 9 01:16:34 WARNING[5719]: channel.c:781 channel_find_locked: Avoided initial deadlock for '0x97f24d8', 10 retries!

Does anyone know what the deadlock message is all about? It is ocurring quite frequently.
This is Asterisk 1.2.14.

Thanks,
Doug







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[davies147 at gmail.com]

FYI, check the changelog for 1.2.14 to 1.2.25 - IIRC, there are a
significant number of deadlock-fixing updates. There is at least one
related to the code where that error message is displayed.

Regards,
Steve

On 1/9/08, Douglas Garstang <dougmig33 at yahoo.com> wrote: