Hi all,
my setup is like this:Asterisk 1.8.2.3 + FreePBX 2.8.1 running on a Debian
squeeze box connected with a SIP trunk to Vega 100 VoIP gateway, which, in
turn, is connected by ISDN PRI to a POTS PBX.
When a call is made to a SIP extension from POTS, it proceeds OK, and the
connections succeeds
I am posting the contents of three files: sip.conf, sip_additional.conf, and
extensions_additional.conf. Tell if anything else is important.
Thanks a lot in advance.
Arik
On Mon, Jan 17, 2011 at 1:28 PM, Andre Magalhaes wrote:
> Can you post your confs.
>
>
> 2011/1/17 A
Hello!
I have compiled Asterisk 1.8.1.1 on my SheevaPlug. It works all right for
me, except for one problem that I have encountered: I can only register a
SIP client (X-Lite in my case) if the secret field of the extension is left
blank. Otherwise it throws a 'bad auth' error.
Does anybody have an
