[Asterisk-Users] Sip Trunk config

Wade Weppler weppler at wwworks-inc.com
Thu Aug 7 08:58:12 MST 2003


I've also run into the "how many lines" problem.

Possibly something similar to incominglimit= and outgoinglimit= in h323.conf
could be implemented in sip.conf?

-wade

> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-
> admin at lists.digium.com] On Behalf Of David Hindmarsh
> Sent: Thursday, August 07, 2003 12:19 AM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] Sip Trunk config
> 
> 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..
> > > > >
> > > > >
> > > > >--
> > > > >______________________________________________
> > > > >http://www.linuxmail.org/
> > > > >Now with e-mail forwarding for only US$5.95/yr
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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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