Re: [asterisk-users] One server, multiple companies
-- Mensaje reenviado --
> From: Eric C. <[EMAIL PROTECTED]>
> To:
> Date: Sun, 9 Dec 2007 19:55:51 -0500
> Subject: [asterisk-users] One server, multiple companies
>
> Hello all,
>
> Just starting to setup asterisk v 1.4.11 and need to run three distinct
> phone systems for three different companies.
> So far, I have inbound lines going to the appropriate dial plan within the
> extensions.conf file. I'm using
>
> exten => _X.,1,NoOp(FROM NUMBER: ${SIP_HEADER(TO):5:10})
>
> to determine which number is being dialed by the caller and then using a
> gotoif to get to correct greeting (correct company).
>
> My question is... lets assume all three companies have extension numbers
> being 2000, 2001 & 2002, how does one separate them?
> Or, lets say the extensions are:
>
> company A --> 2000, 2001,2002
> company B --> 3000, 3001, 3002
> company C --> 4000, 4001, 4002
>
> Since they're on one server with one asterisk process, how can I use
> context correctly so that the user at 4002 cannot get through to the user at
> company A whose extension is 2000 as currently, I can dial 2000 from phone
> 4002.
>
> That's my current problem, how should this be setup? Is my architecture
> correct? Should I be running different processes for each company? Can
> context resolve what I need?
Hi,
You should try DeStar, a management interface for Asterisk:
http://destar.berlios.de/
DeStar supports "Virtual PBXs", then you can install it and take a look at
the dialplan. Sorry for the late answer but I've just read the list
messages.
Bye,
Diego Andrés.
So
Please advise.
>
> thanks,
> Otto
>
>
>
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Re: [asterisk-users] Strange SIP response
Rushowr wrote: Have you run SIP DEBUG PEER 192.168.1.60? It may help...tcpdump is also one of my personal favorites Yes, I have used it. The lines are extracted from a sip debug on the CLI. I'm going to paste more lines: Sip read: SIP/2.0 480 Temporarily Unavailable To: ;tag=e4331437 From: "24307022";tag=as288765a2 Via: SIP/2.0/UDP 172.16.1.3:5060;branch=z9hG4bK73129cd1;received=172.16.1.3 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Content-Length: 0 7 headers, 0 lines -- Got SIP response 480 "Temporarily Unavailable" back from 192.168.1.50 Transmitting: ACK sip:[EMAIL PROTECTED]:6198 SIP/2.0 Via: SIP/2.0/UDP 172.16.1.3:5060;branch=z9hG4bK73129cd1 From: "24307022" ;tag=as288765a2 To: ;tag=e4331437 Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.1.50:6198 -- SIP/EXT25-a454 is circuit-busy == Everyone is busy/congested at this time I have not detected packet losses even. Thanks for your response. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Diego Andres Asenjo G. Sent: Tuesday, August 22, 2006 6:50 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Strange SIP response Hi, I am getting the following message on the CLI: -- Got SIP response 480 "Temporarily Unavailable" back from 192.168.1.60 -- SIP/EXT23-d910 is circuit-busy and the call hangs up. The peer is correctly registered and I'm not getting unavailable messages. I really need help with this error. -- MENSAJE ENVIADO CON WMAIL 1.01 UNIVERSIDAD DEL CAUCA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 480 "Temporarily Unavailable" message
Hi everybody! I have a SIP peer correctly registered on my asterisk server (Status: OK (2ms)). I can call the peer normally from another peers, os th DND is no set. But sometimes I got -- Got SIP response 480 "Temporarily Unavailable" back from 172.16.34.17 -- SIP/XXX-d910 is circuit-busy The peer never loses its registry and there are no packet losses between it and the server. Can someone help me debug and resolve this problem? Thanks a lot. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hangup detection
Hi everybody! Jonathan wrote: > > Hi, > > I'm using a td400p card with an FXO port and asterisk 1.2.1 in South > Korea and asterisk isn't detecting when PSTN callers hangup. > I've gone through all the settings related to hangup detection and none > work. I've tried: > hanguponpolarityswitch=yes > callprogress=yes > busydetect=yes > busycount=6 I'm using asterisk/zaptel 1.0.10 and have the same situation. I'm in Colombia and tried with a lof of loadzone= > > Debug doesn't show reverse polarity events so I'm pretty stuck. > > I've got zaptel configured with a loadzone of US and kewlstart signialling. > > Has anybody had success with these cards/asterisk in South Korea? ¿Or in the world? > > Thanks > JC > > > > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Diego Andrés Asenjo González Universidad del Cauca Ingeniero en Electrónica y Telecomunicaciones signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with SIP register
Hi! I'm registering an asterisk server in a Sysmaster with a SIP account. The registration succeeds and I can establish a call that come from the Sysmaster. After around 80 seconds the Sysmaster sends a BYE SIP message and the call hang up. This does not occur to the hard/soft SIP phones registered in the sysmaster. I debug, but the only info that I can get is the BYE message. Thanks for your suggetions soving the problem. Bye. -- Diego Andrés Asenjo González Universidad del Cauca Ingeniero en Electrónica y Telecomunicaciones signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
