[Asterisk-Users] SIP gateway question

[Christian Hecimovic]

Hi Rich,

Here's how I did it - it's a bit hacky, but it works. It does not require a 
special extension.

In my setup, the 1204 has a static IP of 192.168.1.2; the Asterisk box is 
192.168.1.1.

For outgoing calls:

exten => _NXXNXXXXXX,1,Dial,SIP/${EXTEN}@192.168.1.2

Incoming:

exten => 192.168.1.1,1,Answer
.
.
(etc. - whatever set of answer routines you use)

I've configured the gateway to use Asterisk as its outbound proxy, and also 
enabled port redirection to point to Asterisk.

Problems: provisioning the thing with DHCP is a nightmare. I'm trying to get 
the DHCP payload to include all of the relevant server addresses, and it 
works, except for the outbound proxy parameter. The gateway just won't pick 
it up. It uses everything else perfectly, just not that one thing. I can only 
conclude it's a bug in the gateway firmware.

On long distance calls that start with 0, the gateway always trims the 0 off. 
I have to add it back in manually with another gateway setting. Highly 
annoying.

Overall, it seems reliable but quirky.

Regards,

Christian

On Saturday 31 January 2004 11:59, Rich Adamson wrote:
> Hi Bob,
>
> > >The 1204 then sends "one" more packet to * with both the source and
> > > destination ports one digit greater then what was used for the rtp
> > > session. I'm assuming that's a bug in their code; anyone seen something
> > > like that before?
> >
> > That would be RTCP (RTP + 1)
> >
> > >3. Has anyone played with this box and found any unusual problems, weird
> > >config's, etc?
> >
> > I have several of these boxes in use at a few different sites.
> > Once installed, I have never gone back in and looked at any of them.
> > They just work.
> >
> > I have it running in canreinvite mode and all sip phones running p2p.
> > The poor * box has really no work to do.
>
> I'm trying to figure out how best to bring pstn calls into * using this
> box, and not sure I'm there yet. Since the box doesn't register with *, I'm
> using the Redirect method which effectively causes the 1204 to dial x3094.
>
> What I'd like to do is simply drop that incoming call into the ivr menu
> directly. Any thoughts on how best to do that?
>
> Rich
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users