[Asterisk-Users] Sip Trunk config
David Hindmarsh
dave at lex.net.au
Wed Aug 6 21:18:55 MST 2003
Thanks for that,
I was looking at the extensions.conf, particularly the line in the general
section which is
TRUNK=SIP/???????
Using this method would be easier.
How do you tell asterisk how many lines are available at the gateway
Dave
----- Original Message -----
From: "Martin Pycko" <martinp at digium.com>
To: <asterisk-users at lists.digium.com>
Sent: Thursday, August 07, 2003 12:34 PM
Subject: Re: [Asterisk-Users] Sip Trunk config
> 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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> > > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > > 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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> > >
> >
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>
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