On Monday 13 March 2006 20:47, Dave Hope wrote:> Hello all,
>
> With some help from people in #asterisk on freenode, I've managed to
get
> incoming SIP calls working.
>
> Outgoing calls however are however a different matter. My whole working
> (incoming calls only) SIPgate configuration can be found here. [1]
>
> When I uncommon what's in there, nothing works. There doesn't
appear to
> be any useful error being logged , even when debug is enabled for
> console and file logs.
>
> If anyone could take a look and show me what needs adding in order for
> outgoing calls to work, that would be superb!
>
> My long term goal is to get asterisk running at home, and then persuade
> the boss to ditch the Avaya setup we have at the office. But since I'd
> likely be the one implementing it, I want to try and get something
> working before I commit myself :)
>
> Thanks!,
>
> Dave.
>
> [1] http://files.davehope.co.uk/home.tar
Hi!
I think it was a bad idea to make people download a tar just to help you...
anyway, I did it and my first piece of advice would be, that you should
implement something where the user should dial a 0 for an outside line. So in
the dialplan you would have something like this:
exten => _0X.,1,Dial(SIP/${EXTEN:1}@Sipgate-out,60)
exten => _0X.,2,Hangup
So now you have to dial a 0, then the number you want to call and so it goes
out over the sipgate account... What's different here is that it's _0X.,
(see
the dot)? That should make a little difference.
Hope it helps!
Christoph
>
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