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" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> 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: [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 > >> > > >> > >> _______________________________________________ > >> 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 > > _______________________________________________ > 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
