Re: [asterisk-users] One server, multiple companies
-- Mensaje reenviado --
From: Eric C. [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
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: sip:[EMAIL PROTECTED]:6198;tag=e4331437 From: 24307022sip:[EMAIL PROTECTED];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 sip:[EMAIL PROTECTED];tag=as288765a2 To: sip:[EMAIL PROTECTED]:6198;tag=e4331437 Contact: sip:[EMAIL PROTECTED] 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
