[Asterisk-Users] Sip Trunk config
Ricardo Villa
ricvil at telesip.net
Thu Aug 7 15:12:06 MST 2003
Hi Todd,
This limit on outbound calls looks interesting. Can you provide an example?
I have not used db routines before.
Thanks,
Ricardo Villa
http://www.telesip.net
----- Original Message -----
From: "John Todd" <jtodd at loligo.com>
To: <asterisk-users at lists.digium.com>
Sent: Thursday, August 07, 2003 2:51 PM
Subject: RE: [Asterisk-Users] Sip Trunk config
> And to answer Wade's question: to limit outbound calls on a
> particular path, you'd use a local db set routine. In other words,
> every time a call is created to that particular SIP peer, you'd add 1
> to the counter, and every time a call was hung up out of that pool,
> you'd subtract one.
>
> JT
>
>
> At 3:30 PM -0400 8/7/03, Patrick wrote:
> >
> >incominglimit is already implemented for SIP. Just specify under the
> >endpoint how many incoming connections are allowed.
> >
> >For example,
> >
> >[cisco]
> >type=friend
> >username=cisco
> >secret=blah
> >nat=yes ; This phone may be natted
> >host=dynamic
> >canreinvite=no ; Cisco poops on reinvite sometimes
> >qualify=200 ; Qualify peer is no more than 200ms away
> >defaultip=192.168.0.4
> >incominglimit=20 ; set limit to 20 voice channels
> >
> >
> >setting the limit to 0 (incominglimit=0) is unlimited.
> >
> >to view the current lines in use --- sip show inuse from the cli.
> >
> >
> >Patrick
> >
> >
> >> 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
> >> > > > >
> >> > > > >Powered by Outblaze
> > > > > > >_______________________________________________
> >> > > > >Asterisk-Users mailing list
> >> > > > >Asterisk-Users at lists.digium.com
> >> > > > >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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