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: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of David Hindmarsh Sent: Thursday, August 07, 2003 12:19 AM To: [EMAIL PROTECTED] 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" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, August 07, 2003 12:34 PM Subject: Re: [Asterisk-Users] Sip Trunk config
> exten => _9X.,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED] > > 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" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > 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 ." <[EMAIL PROTECTED]> 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: [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> > > > >> Phone: +61 414 365 466 > > > >> Jabber: [EMAIL PROTECTED] > > > >> > > > > > > > >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 > > > >[EMAIL PROTECTED] > > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > Regards, > > > > > > Jamie Carl > > > Jazz Inc. > > > Email: [EMAIL PROTECTED] > > > Web: www.jazz-inc.net > > > Phone: +61-414-365-466 > > > Jabber: [EMAIL PROTECTED] > > > _______________________________________________ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users >
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