Dave Donovan wrote:
On Tue, Apr 22, 2008 at 6:14 PM, Rob Sutherland <[EMAIL PROTECTED]> wrote:
Usually the first place to look when troubleshooting is the asterisk
Console. You can get that by logging in to your box and typing "asterisk
-r". Trixbox is a blessing and a curse in this regard, it puts out lots and
lots of detail when it's processing calls. From my memory, you could easily
see a hundred lines on the console for just one call. It's a bit of a
needle in a haystack, but your needle is probably in there.
Yes, your memory is correct :-) I guess my main question is 'What do
these needles look like?' and/or 'Where do most of them come from? I'm
finding the console fairly useless, as you say, because of the
volume of messages, likewise cating the log. It looks like the first
debugging technique I need to master is how to control the debug output.
Freepbx dumps so many messages I can't even complete the debug related
console keyins :-(
If you give us some more details on your problem, someone here might be able
to point you in the right direction. I'm not sure if I'm reading that
right, User A calls User B and the call completes but A can't hear B? Can B
hear A?
No, just dead air, A to B and B to A. Only on an attempt to connect from
one extension to another. Both A and B can call out and receive calls
from outside.
Also, some info about the types of phones can help. How is the network
configured? Are both parties on the same LAN or is there routing going on?
Are these SIP phones? Which models? Is it always one model or unit that
can't hear? People on this list have worked with just about every kind of
equipment I can imagine and some kinds that I can't. Often you'll find that
if you can describe the circumstances specifically, someone will say "Oh
yeah, I had that problem, here's how I solved it...".
We're using Polycom 320 SIP phones - I'm using an X-Lite SIP softphone
as well, same behaviour. Everything is on the same LAN, no routing
involved.
Rob
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