[asterisk-dev] SIP register expiry timer
Joshua Foshee
Joshua.Foshee at LogixCom.com
Tue May 4 12:10:24 CDT 2010
I am using the CLI to get the debug.
I have the verbose set to 255 and the sip debug on
Josh
-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Nick Lewis
Sent: Tuesday, May 04, 2010 11:14 AM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] SIP register expiry timer
>Not withstanding the other issues you could enable debug on the cli to
>see why the 200 OK is not being matched to the REGISTER (causing the
>retransmission). This could be because find_call() is not able to match
>the Call-ID or the from-tag or it could be due to a min-expiry problem.
>Either way, the cli debug should elucidate
Sorry Josh I did not read before write
I think that the cli is more use than the log as find_call uses
ast_debug (rather than
ast_log) for its output which I think may only go to the cli
-- N_L
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