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@lists.digium.com [mailto:asterisk-users-
>> admin@lists.digium.com] On Behalf Of David Hindmarsh
>> Sent: Thursday, August 07, 2003 12:19 AM
>> To: asterisk-users@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@digium.com>
>> To: <asterisk-users@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@jazz-inc.net>
>> > > To: <asterisk-users@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@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@jazz-inc.net
<mailto:egeek@jazz-inc.net>
>> > > > >> Phone: +61 414 365 466
>> > > > >> Jabber: jazz@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@lists.digium.com
>> > > >
>http://lists.digium.com/mailman/listinfo/asterisk-users
>> > > >
>> > > > Regards,
>> > > >
>> > > > Jamie Carl
>> > > > Jazz Inc.
>> > > > Email: me@jazz-inc.net
>> > > > Web: www.jazz-inc.net
>> > > > Phone: +61-414-365-466
>> > > > Jabber: jazz@netmindz.net
>> > > > _______________________________________________
>> > > > Asterisk-Users mailing list
>> > > > Asterisk-Users@lists.digium.com
>> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
>> > > >
>> > >
>> > > _______________________________________________
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>>
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