[Asterisk-Users] Outgoing calls via Sipgate

[Dave Hope]

On Mon, 2006-03-13 at 21:13 +0100, Christoph Eicke wrote:
> 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... 

Hi Chris,

Apologies for that. Didn't know that wasn't the done thing on this ML,
have tar xf'd the files to:

	http://files.davehope.co.uk/asterisk/

> 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

When adding that and un-commenting my stuff in sip.conf I was still
unable to dial out, but could no longer dial in. The log for incoming
calls (now broken) shows:

        chan_sip.c:7329 handle_request: Check for res for 9448153
        chan_sip:1592 update_user_counter: 9448153 is not a local user
        chan_sip:1592 update_user_counter: 9448153 is not a local user
        chan_sip.c:840 __sip_ack: Stopping retransmission on
        '79e17a915abdd68f015545db215cd7cf at gw02.sipgate.net' of Response
        102: Found



> 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!

Thanks for the help!,

Dave

> Christoph
> 
> 
> 
> >
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