[asterisk-users] SIP trunk between two Asterisk servers [SOLVED]

[Vieri]

--- On Fri, 5/14/10, Philipp von Klitzing <klitzing at pool.informatik.rwth-aachen.de> wrote:

> You were probably caught be the fact that you are using
> extension numbers 
> also as SIP user names for your phones (here: 3666). This
> is not a good 
> thing to do, better use an alphanumeric username or the
> phone's MAC 
> address etc.

Is there more info on this?
I mean, why is it "bad", apart from the security implication.

> As for your IAX sound quality issue: I have seen that
> before as well, and 
> switched to SIP (as others did). My guess is that it will
> probably go 
> away if you use Asterisk 1.4 on both sides, though.

It went away even with 1.2 but I needed to set trunk=no.
Probably a jitter buffer issue on my system(s).

> SIP DEBUG on the receiving Asterisk gives you a hint which
> peer was found 
> if matching is done on the IP address, the text is
> somethint like "Found 
> peer ..." or "Found no matching peer or user for w.x.y.z"

Tnanks for the info Philipp.
I'll try to further debug my SIP messages.

Vieri