[Asterisk-Users] Sip Trunk config

[Martin Pycko]

exten => _9X.,1,Dial,SIP/${EXTEN:1}@ip_of_the_gateway

regards
Martin

On Thu, 7 Aug 2003, David Hindmarsh wrote:

> Hi
>
> Is it possible to use a sip gateway as a trunk.
>
> If so,  how would I do this
>
> David Hindmarsh
>
> ----- Original Message -----
> From: "Jamie Carl" <geek at jazz-inc.net>
> To: <asterisk-users at lists.digium.com>
> Sent: Thursday, August 07, 2003 12:14 PM
> Subject: Re: [Asterisk-Users] X-Lite <-> Snom200
>
>
> > Yes, over a LAN.  It does it with both g.711 and GSM which
> > both used to work.  Havn't had a chance to have a REAL
> > good look into it though.
> >
> > J
> >
> > On Wed, 06 Aug 2003 14:33:47 +0000
> >   "WipeOut ." <wipeout at linuxmail.org> wrote:
> > >*This message was transferred with a trial version of
> > >CommuniGate(tm) Pro*
> > >> *This message was transferred with a trial version of
> > >>CommuniGate(tm) Pro*
> > >> Dunno what I'm doing wrong here but I just did an
> > >>upgrade to the latest
> > >> version and now I get no audio at all!
> > >> I havn't changed a single thing.  Is there anything
> > >>special I need to do
> > >> to get this to work again?
> > >>
> > >> I get a quick 'chirp' of audio, which you can tell is
> > >>what I'm
> > >> connecting to, (ie MOH), but then nothing.
> > >>
> > >>
> > >> Regards,
> > >>
> > >> Jamie Carl
> > >> Email:  geek at jazz-inc.net <mailto:egeek at jazz-inc.net>
> > >> Phone:  +61 414 365 466
> > >> Jabber: jazz at netmindz.net
> > >>
> > >
> > >Are you connecting to * over a LAN?? I have experienced
> > >the "chirp" when the phone was trying to use G.711 over a
> > >dial up link so there was not enough bandwidth..
> > >
> > >
> > >--
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> >
> > Regards,
> >
> > Jamie Carl
> > Jazz Inc.
> > Email:  me at jazz-inc.net
> > Web:    www.jazz-inc.net
> > Phone:  +61-414-365-466
> > Jabber: jazz at netmindz.net
> > _______________________________________________
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> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
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