[Asterisk-Users] Sip Trunk config

[John Todd]

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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