[asterisk-users] dial_exec_full problems with TDM400

[Jason Morgan]

Hi AJ,

Surely this is a really bad bug if unconnected SIP devices ( a very likely
occurance ) can take out trunk lines.

Anyway what you say is true, there are several sip phones defined and not
all are physically present all the time.

I'll remove definitions of all sip phones for a while and see what that
does.

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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