[asterisk-users] dial_exec_full problems with TDM400 - getting critical.

Jason Morgan jason.morgan at vpnsolutions.uk.com
Sun Aug 22 11:19:45 CDT 2010


Hi,

I thought you'd cracked it, I simply turned off all sip by removing the
sip.conf
but after a few more days it did the same.

I've set logging permanently on again.

Any other suggestions?

Cheers,
Jason.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of A J Stiles
Sent: 17 August 2010 10:36
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] dial_exec_full problems with TDM400


On Tuesday 17 Aug 2010, Jason Morgan wrote:
> Hi,
>
> I was running asterisk 1.4, but recently upgraded to 1.6 (for fax support)
> at the same
> time as moving from Ubuntu hardy to
>
> I have a single TDM400P rev I with two fxo and two fxs modules, these were
> working perfectly for years
> on Asterisk 1.4 using Zaptel drivers with Oslec.
>
> Now I've moved to 1.6 so I am using Dahdi.  Distribution is stock ubuntu
> package.
>
> After several hours (perhaps 24 or so, not nailed it down precisely)
> incoming
> calls are not answered and outgoing calls get dial_exec_full.
>
> Incoming calls are reported to either A:just ring and ring, or B:get an
> engaged tone.
>
> Strangely when this happens asterisk DOES see the incoming call in
> situation A, but fails
> to answer.
>
> What tests can I do to resolve this as it is very inconvenient as we are
> missing a lot of calls?

Have you got any extensions defined that aren't physically connected to
anything?

I replaced an ancient Asterisk version with a bang-up-to-date 1.6 version I
built myself, and was getting similar symptoms to what you describe.  It
seemed not to be freeing up channels it was trying to associate with
non-existent devices.

I made sure that every entry in sip.conf had a corresponding phone plugged
in
somewhere, then went through the dialplan and removed all references to
anything that wasn't mentioned in sip.conf.  (And there were a few.)  It
seems to have stayed up since then .....

--
AJS

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