Someone? As * is used so extensively with SIP I must've made a _glaring_
mistake in my config (!)
/Rob
Robert Bielik skrev:
Tarek Sawah skrev:
you need to post you SIP.conf and your Extensions.conf so someone can
have a look at them and see if there is anything missing
what are the
Lacking any response I tried to set insecure=invite on both sides. And lo and
behold, the call
gets through.
Now, is this good or bad?
/R
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Discussion
Subject: Re: [asterisk-users] SIP interconnection problem
Lacking any response I tried to set insecure=invite on both sides. And lo
and behold, the call
gets through.
Now, is this good or bad?
/R
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Tarek Sawah skrev:
you need to post you SIP.conf and your Extensions.conf so someone can
have a look at them and see if there is anything missing
what are the contexts you are using with your peers?
what is the dial plan triggered when calling your destination number?
Machine 1
Ooops.. forgot. The versions of * are:
Machine 1: 1.6.1.4
Machine 2: 1.6.0.5
/Rob
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Hi all,
I've setup two * servers which are SIP interconnected ala osaka/toronto from
the * book (before anyone sugggests using
IAX instead, no, I NEED to have them SIP interconnected for verification/test
purposes). Then I have a
Zoiper connected to one of them via IAX (so that * will not
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
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Date: Sun, 25 Oct 2009 15:19:28 +0100
From: robert.bie...@xponaut.se
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] SIP interconnection problem
Hi all,
I've setup two * servers which are SIP
