[Asterisk-Users] Sip Trunk config / Least Cost Routing
John Todd
jtodd at loligo.com
Thu Aug 7 13:48:46 MST 2003
See answers in-line.
At 4:14 PM -0400 8/7/03, Wade Weppler wrote:
>From: "Wade Weppler" <weppler at wwworks-inc.com>
>To: <asterisk-users at lists.digium.com>
>Subject: RE: [Asterisk-Users] Sip Trunk config / Least Cost Routing
>Reply-To: asterisk-users at lists.digium.com
>Date: Thu, 7 Aug 2003 16:14:51 -0400
>
>Ah, good idea! I assume even a global variable could be used instead of
>using db routines...
>
>Where this doesn't work so well is when trying to implement
>least-cost-routing using local calling areas spread over satellite offices.
>Here's an example:
>
>Office A has 4 Telco lines.
>
>Office B has 4 Telco lines.
>
>Office A and Office B have 8 station sets each.
>
>Office A and Office B both have Asterisk boxes.
>
>Office A and Office B are long distance calls away from each other, so they
>use IAX for interoffice calls, and would also like to utilize VoIP to extend
>their local calling area.
>
>If an employee from Office A wants to make a call to someone in Office B's
>local calling area, the system will need to follow the following logic:
>
>1) Is there a telco line available in Office B?
>2) No? Use a local line in Office A and make a long distance call.
>3) Yes? Place the call through a local line in Office B.
>4) Worst case, all lines are busy. Let the user know.
>
>Bottom line, the call has to go through without any intervention from the
>user, but try the cheapest method first.
>
>We're already written an AGI module to handle the call routing (ie. which
>numbers are locally available from each Office), but I'd like to be able to
>handle line availability as well.
Why use an AGI? This seems to be easily done with the dialplan,
unless I'm missing some additional sophistication that you're not
mentioning.
>Any idea how this could be done?
TRIP (RFC2871 and RFC3219) Not implemented in Asterisk yet -
looking for programmers. See my posts to the -dev list last month.
This is probably overkill for a two office situation, but imagine you
have three hundred offices...
JT
>-wade
>
>
>
>
>> -----Original Message-----
>> From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-
>> admin at lists.digium.com] On Behalf Of John Todd
>> Sent: Thursday, August 07, 2003 3:51 PM
>> To: asterisk-users at lists.digium.com
>> 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
>> >> > > > _______________________________________________
>> >> > > > Asterisk-Users mailing list
>> >> > > > Asterisk-Users at lists.digium.com
>> >> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
>> >> > > >
>> >> > >
>> >> > > _______________________________________________
>> >> > > Asterisk-Users mailing list
>> >> > > Asterisk-Users at lists.digium.com
>> >> > > http://lists.digium.com/mailman/listinfo/asterisk-users
>> >> > >
>> >> >
>> >> > _______________________________________________
>> >> > Asterisk-Users mailing list
>> >> > Asterisk-Users at lists.digium.com
>> >> > http://lists.digium.com/mailman/listinfo/asterisk-users
>> >> >
>> >>
>> >> _______________________________________________
>> >> Asterisk-Users mailing list
>> >> Asterisk-Users at lists.digium.com
>> >> http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
>> >_______________________________________________
> > >Asterisk-Users mailing list
>> >Asterisk-Users at lists.digium.com
>> >http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
>> >_______________________________________________
>> >Asterisk-Users mailing list
>> >Asterisk-Users at lists.digium.com
>> >http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users at lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
More information about the asterisk-users
mailing list