[Asterisk-Users] Sip Trunk config / Least Cost Routing
Wade Weppler
weppler at wwworks-inc.com
Thu Aug 7 13:14:51 MST 2003
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.
Any idea how this could be done?
-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
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> >> > > >
> >> > > > 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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> >> > > >
> >> > >
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