[Asterisk-Users] Sip Trunk config

[David Hindmarsh]

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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