[Asterisk-Users] SIP Absolute Timeout

[Wes Marderness]

Thanks, I got the latest CVS and the timeout worked as it should. I did get
an error message in the console.

[warning](15375): rtp.c:1204 ast_rtp_bridge: codec0=524556 is not
codec1=524558: can not native bridge

I was using G729b codec for the call. I didn't notice any problems with the
call everything worked as it should.

thanks again,
Wes

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Brian West
Sent: Friday, January 23, 2004 10:28 AM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] SIP Absolute Timeout


I think this has been fixed since 0.5.0 their was a problem with timeout's
and native bridges.  Might wanna update.

bkw

On Fri, 23 Jan 2004, Wes Marderness wrote:

> Hi All,
>
> I've been having a hard time getting the AbsoluteTimeout function to work.
> Is this Function working in for SIP? I've search all the messages in the
> news letters and tried what was suggested and still have not gotten it to
> work. Below is a portion of my extensions.conf. I've also been running
these
> test on ver 0.5.0
>
> exten => _X.,1,Absolutetimeout(20)
> exten => _X.,2,dial(SIP/${EXTEN}@SIPOUT#1)
>
> exten => T,1,BackGround(tt-weasels)
> exten => T,2,Hangup()
>
> Thanks ahead of time for any help / suggestions.
> Wes Marderness
>
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