RE: [Asterisk-Users] Realtime Voicemail
-Original Message- From: Douglas Garstang Sent: Tuesday, June 27, 2006 11:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Realtime Voicemail I'm noticing that the documentation on the voip wiki for voicemail and realtime voicemail hasn't kept up with reality. I just created a column called maxmsg in my table. I set it to 1 for the user. I can leave more than once voicemail message. Why? Weird. Maxmsg suddenly worked on the next call. I tried setting maxlogins for the user to 1, and it's letting me put the wrong pin in 3 times before disconnecting me. What am I missing here? Are the supported options documented somewhere, that matches up with what's really in the code? Doug. ___ --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] Realtime Voicemail
-Original Message- From: Michiel van Baak [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 27, 2006 12:21 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Realtime Voicemail On 12:13, Tue 27 Jun 06, Douglas Garstang wrote: -Original Message- From: Douglas Garstang Sent: Tuesday, June 27, 2006 11:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Realtime Voicemail I'm noticing that the documentation on the voip wiki for voicemail and realtime voicemail hasn't kept up with reality. I just created a column called maxmsg in my table. I set it to 1 for the user. I can leave more than once voicemail message. Why? Weird. Maxmsg suddenly worked on the next call. I tried setting maxlogins for the user to 1, and it's letting me put the wrong pin in 3 times before disconnecting me. What am I missing here? Are the supported options documented somewhere, that matches up with what's really in the code? Do you cache realtime stuff ? If so, that would explain it I wasn't aware that realtime voicemail supported caching. I knew sip.conf did, but voicemail? How does that work? I just tried setting 'format' and 'sendvoicemail' in the users database row. No effect. BUT... maxmsg DOES work... I don't understand. Doug. ___ --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] Realtime Voicemail
-Original Message- From: Douglas Garstang Sent: Tuesday, June 27, 2006 12:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Realtime Voicemail -Original Message- From: Douglas Garstang Sent: Tuesday, June 27, 2006 11:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Realtime Voicemail I'm noticing that the documentation on the voip wiki for voicemail and realtime voicemail hasn't kept up with reality. I just created a column called maxmsg in my table. I set it to 1 for the user. I can leave more than once voicemail message. Why? Weird. Maxmsg suddenly worked on the next call. I tried setting maxlogins for the user to 1, and it's letting me put the wrong pin in 3 times before disconnecting me. What am I missing here? Are the supported options documented somewhere, that matches up with what's really in the code? Oh man, this is some freaky stuff. I commented out 'format=wav49|gsm|wav' in voicemail.conf and did a reload. I set the format field to the user to 'gsm'. And... Asterisk record a wav file. Huh? Doug. ___ --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] Polycom Buddies in 1.6.6
I've
never seen that problem, and I've only ever used 1.2+ with Polycom and
buddies.
-Original Message-From: Ryan Stark
[mailto:[EMAIL PROTECTED]Sent: Tuesday, June 27, 2006 12:31
PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [Asterisk-Users] Polycom Buddies in
1.6.6So I've got a 601 (1.6.6) with the side car, and the
buddy watch seems to be working but it updates the statuses unreliably.
When I do a sip show subscriptions in asterisk it lists my phone 12 times and
at the bottom it says "0 active SIP subscriptions(s)" I've got an older
CVS-HEAD build, pre 1.2, do you think my problems are polycom or asterisk
based?-Ryan
On 6/19/06, Kevin P.
Fleming [EMAIL PROTECTED]
wrote:
-
Douglas Garstang
[EMAIL PROTECTED] wrote: Polycom released their
SIP software version 1.6.6 for their phones recently. I was under
the impression that this release fixed a previous limitation where
the phones would only watch 7 buddies, ie send 7 sip subscriptions
to Asterisk. I have configured a phone directory to watch 30 or so
appearances, and it still seems to only be sending 7 subscriptions
to Asterisk. Has anyone else got this to work? Yes,
it works on the Polycom 601 on my desk. However, the release notes say that
the restriction was only removed for the IP600 and IP601; if you are using
an IP300/1, IP500/1 or IP430 than the 7 buddy limit will still be in effect.
--Kevin P. FlemingSenior Software EngineerDigium,
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[Asterisk-Users] Realtime Voicemail Broken?
What's
up with realtime voicemail? I have been going thtough and testing each feature
that can be set as a column in the db, one by one.
Some
do work, and some don't.
Here's
some I have found that do work:
delete
envelope
maxmsg
review
saycid
and
here's some that simply don't work:
attach
(emails sentif there is something in the email field)
maxsilence (docs say the default is 0/off, but the default is
10s)
maxmessage
minmessage
maxlogins (how hard can this be?)
pbxskip
Has
anyone got any idea on this?
Doug.
-Original Message-From: Andrew Nowrot
[mailto:[EMAIL PROTECTED]Sent: Tuesday, June 27, 2006 2:12
PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [Asterisk-Users] Call length
limitationOn 6/27/06, William
Piper [EMAIL PROTECTED] wrote:
Well, It was worth a shot.
Perhaps doing a some variation of the HANGUPCAUSE variable.
http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+hangupcause
exten = x,2,Dial(Sip/|30|gL(6:3:1))
exten = x,3,GoToIf($["${HANGUPCAUSE}" != "1"]?4:10)exten
= x,4,DeadAGI()
exten = x,10,hangup()I
will do that first thing in the morning (now it's getting late) and of course
send the results to the list. Cheers
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RE: [Asterisk-Users] Asterisk Startups
Yeah, that's what I like about Oz. Everyone knows everyone... miss you guys too! -Original Message- From: Rob Thomas [mailto:[EMAIL PROTECTED] Sent: Monday, June 26, 2006 2:58 AM To: asterisk-users Subject: RE: [Asterisk-Users] Asterisk Startups Well now would be a great time to come back, Doug! We miss you! 8) --Rob -Original Message- From: Douglas Garstang [mailto:[EMAIL PROTECTED] Sent: Monday, 26 June 2006 3:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Startups Paul, D'oh. The fact I left Sydney 5 years ago for the US might be a teeny complication. :P Doug. -Original Message- From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Sun 6/25/2006 11:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Asterisk Startups Douglas Garstang wrote: Does anyone know of any startups using Asterisk? What about established companies? Ones that are hiring would be nice :) Doug. -- -- ___ --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 We are always looking for good people - here in Melbourne. PaulH -- Paul Hales Technical Manager AsteriskIT www.asteriskit.com.au bus: 03 8320 8100 mob: 0434 673 529 ___ --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 ___ --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] '500 Internal Server' Error on SIP NOTIFY
Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? I called Polycom tech support, who where utterly useless. Of course Polycom won't officially support it anyway, as they only support Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for quite some time. We have about 35 phones and it's happening on most (also on the few running SIP software 1.6.6). SIP Software version: 1.6.3.0067 BootROM version: 2.6.2.0032 Reliably Transmitting (no NAT) to xxx.187.128.95:5060: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport From: sip:[EMAIL PROTECTED];tag=as6fd80d1b To: Front Desk sip:[EMAIL PROTECTED];tag=3B576862-120A3007 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 114 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: presence Content-Type: application/xpidf+xml Subscription-State: active Content-Length: 371 ?xml version=1.0? !DOCTYPE presence PUBLIC -//IETF//DTD RFC XPIDF 1.0//EN xpidf.dtd presence presentity uri=sip:[EMAIL PROTECTED];method=SUBSCRIBE / atom id=2944026 address uri=sip:[EMAIL PROTECTED];user=ip priority=0.80 status status=open / msnsubstatus substatus=online / /address /atom /presence -- SIP read from xxx.187.128.95:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport From: sip:[EMAIL PROTECTED];tag=as6fd80d1b To: Front Desk sip:[EMAIL PROTECTED];tag=3B576862-120A3007 CSeq: 114 NOTIFY Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Event: presence User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Content-Length: 0 Doug. ___ --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] '500 Internal Server' Error on SIP NOTIFY
-Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Monday, June 26, 2006 11:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] '500 Internal Server' Error on SIP NOTIFY Douglas Garstang wrote: Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? Yes, for quite a while. Happens for us, when you do a transfer via the Polycom's transfer button. Doesn't seem to cause any problems though. It's bloody annoying though, especially for those type-A's that don't like to see the console cluttered up with junk. :) Doug ___ --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] AGI script can not print out error message toconsole
-Original Message- From: Moises Silva [mailto:[EMAIL PROTECTED] Sent: Monday, June 26, 2006 2:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AGI script can not print out error message toconsole what do you mean by could not print out message to stderr??? Try being more descriptive about your problem. Error messages, how have you tried etc. On 6/26/06, Zichao Wu [EMAIL PROTECTED] wrote: Hi, guys, I used /usr/src/asterisk/agi/eagi-test.c script to test AGI API, but that script could not print out message to stderr. any ideas? He may be referring to the fact that when you run asterisk in non-console mode, stderr goes nowhere (not even /var/log/asterisk/messages). Considering that in a production environment, your going to want to run it like this, it means that if, say, an AGI script encounters a syntax error, you can't see what the problem was, unless you shut asterisk run, re-run it in console mode, debug, and restart it again. Not very convenient! Doug. ___ --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] Asterisk Startups
Does anyone know of any startups using Asterisk? What about established companies? Ones that are hiring would be nice :) Doug. ___ --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] Asterisk Startups
Paul, D'oh. The fact I left Sydney 5 years ago for the US might be a teeny complication. :P Doug. -Original Message- From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Sun 6/25/2006 11:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Asterisk Startups Douglas Garstang wrote: Does anyone know of any startups using Asterisk? What about established companies? Ones that are hiring would be nice :) Doug. ___ --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 We are always looking for good people - here in Melbourne. PaulH -- Paul Hales Technical Manager AsteriskIT www.asteriskit.com.au bus: 03 8320 8100 mob: 0434 673 529 ___ --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
RE: [Asterisk-Users] News: Asterisk VOIP Jobs Site - Revision 3.0 up!
Well, I hope some more jobs get posted. I took a look tonight, and there was 2 there. -Original Message- From: Matt Gibson [mailto:[EMAIL PROTECTED] Sent: Sun 6/25/2006 11:25 PM To: asterisk-users@lists.digium.com Cc: Subject: [Asterisk-Users] News: Asterisk VOIP Jobs Site - Revision 3.0 up! To all Employable Asterisk Professionals, We are very pleased to announce the unveiling of the newest incarnation of the popular, OpenSource VOIP jobs forum at http://www.asterisk-jobs.com. We at Asterisk-Jobs.com appologize for the inactivity for the past while. It had come to our attention that the software running the job board was unsecure and allowed for multiple Vulnerabilities. The site also previously required the use of PHP's register_globals, which is of course less than desirable. We here at Asterisk Jobs take your information and security seriously. Thus, we decided to lay low for a while while the site was upgraded to be the secure, robust and working system that you - our users expect. And now without furthur ado, is the unveiling of Asterisk Jobs Ver 3.0. This site includes multiple fixes to the html, css and back end code. The resume preview is also functioning now, and the ability to upload pdf and png has been added, furthur allowing for easier use. This free site allows you to post your credentials and search for jobs located near you, or for jobs abroad if you wish to travel. We cater to all segments of the Asterisk OpenSource VOIP Employment market, from small contracts to full system configuration and deployment opportunities. There is no limit to what you can find at Asterisk Jobs. We have recently introduced a free plan for the employers and contractors who use the site. The intent is to get more jobs posted from some of the smaller shops or contractors who do not or cannot afford a paying plan. Cheaper for the employer, more job opportunities for the users, win win! What are you waiting for? Click the link and find that job you've always been dreaming about, or just check out the market to keep yourself ahead of the curve at http://www.asterisk-jobs.com or http://www.asteriskjobs.com Thanks for your support! The Asterisk Jobs Team [EMAIL PROTECTED] ___ --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 winmail.dat___ --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] Caller ID Matching in extensions.conf
I'm running 1.2.9.1, and I can't get caller id dialplan matching to work.
When calling from 9220370 to 1234, the following does not match.
exten = 9220370/1234,1,NoOp(${CALLERIDNUM})
exten = 9220370/1234,2,Answer
exten = 9220370/1234,3,Playback(tt-weasels)
However, when calling from 9220370 to 1234, this DOES match.
exten = 1234,1,NoOp(${CALLERIDNUM})
exten = 1234,2,Answer
exten = 1234,3,Playback(tt-weasels)
You can also see from the console output that the caller id IS 9220370.
-- Executing NoOp(SIP/9220370-7a11, 9220370) in new stack
-- Executing Answer(SIP/9220370-7a11, ) in new stack
-- Executing Playback(SIP/9220370-7a11, tt-weasels) in new stack
-- Playing 'tt-weasels' (language 'en')
What am I missing here?
Oh, this also doesn't match EVER... so I am wondering if there's a problem with
dialplan caller id matching in 1.2.9.1?
exten = _X./1234,1,NoOp(${CALLERIDNUM})
exten = _X./1234,2,Answer
exten = _x./1234,3,Playback(tt-weasels)
Doug.
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RE: [Asterisk-Users] Caller ID Matching in extensions.conf
Oops. You are correct. My bad.
-Original Message-
From: Kevin Collins [mailto:[EMAIL PROTECTED]
Sent: Friday, June 23, 2006 1:35 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Caller ID Matching in extensions.conf
Callerid should be the second argument based on what works for me
Kevin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Friday, June 23, 2006 3:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Caller ID Matching in extensions.conf
I'm running 1.2.9.1, and I can't get caller id dialplan
matching to work.
When calling from 9220370 to 1234, the following does not match.
exten = 9220370/1234,1,NoOp(${CALLERIDNUM})
exten = 9220370/1234,2,Answer
exten = 9220370/1234,3,Playback(tt-weasels)
However, when calling from 9220370 to 1234, this DOES match.
exten = 1234,1,NoOp(${CALLERIDNUM})
exten = 1234,2,Answer
exten = 1234,3,Playback(tt-weasels)
You can also see from the console output that the caller id
IS 9220370.
-- Executing NoOp(SIP/9220370-7a11, 9220370) in new stack
-- Executing Answer(SIP/9220370-7a11, ) in new stack
-- Executing Playback(SIP/9220370-7a11, tt-weasels)
in new stack
-- Playing 'tt-weasels' (language 'en')
What am I missing here?
Oh, this also doesn't match EVER... so I am wondering if
there's a problem
with dialplan caller id matching in 1.2.9.1?
exten = _X./1234,1,NoOp(${CALLERIDNUM}) exten =
_X./1234,2,Answer exten =
_x./1234,3,Playback(tt-weasels)
Doug.
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[Asterisk-Users] Showing Current Calls
Can someone recommend the best way to view current calls in progress on the Asterisk console? Neither the 'show channels' or 'sip show channels' commands are easy to read. hestia*CLI show channelsChannel Location State Application(Data) SIP/2944093-f9e2 (None) Up Bridged Call(SIP/2944079-e7f2)SIP/2944079-e7f2 [EMAIL PROTECTED]:2 Up Dial(SIP/2944093|36|tr) 2 active channels1 active call hestia*CLI hestia*CLI sip show channelsPeer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Messagexxx.yyy.128.115 (None) e77bba33-cc 00101/02261 unkn No Rx: REGISTERxxx.yyy.128.110 (None) 739f4603-e8 00101/00778 unkn No Rx: REGISTERxxx.yyy.128.86 (None) 56caad3a-eb 00101/01046 unkn No Rx: REGISTERxxx.yyy.128.115 (None) 91ea0410-60 00101/02262 unkn No Rx: REGISTERxxx.yyy.128.86 (None) 488801e-105 00101/01046 unkn No Rx: REGISTERxxx.yyy.128.86 (None) c3b27274-ef 00101/01194 unkn No Rx: REGISTERxxx.yyy.128.77 2944093 2405f1ef74d 00102/0 ulaw No Tx: ACKxxx.yyy.128.83 2944079 cf1722ef-cc 00101/2 ulaw No Rx: ACK Doug. ___ --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] Showing Current Calls
Using this as an example: hestia*CLI show channels Channel Location State Application(Data) SIP/2944093-f9e2 (None) Up BridgedCall(SIP/2944079-e7f2) SIP/2944079-e7f2 [EMAIL PROTECTED]:2 Up Dial(SIP/2944093|36|tr) Why does the first line show bridged call, while the second does not? Why is the Location for the first line (None)? -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Thursday, June 22, 2006 1:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Showing Current Calls Whats wrong with show channels? On 6/22/06, Douglas Garstang [EMAIL PROTECTED] wrote: Can someone recommend the best way to view current calls in progress on the Asterisk console? Neither the 'show channels' or 'sip show channels' commands are easy to read. hestia*CLI show channels Channel Location State Application(Data) SIP/2944093-f9e2 (None) Up Bridged Call(SIP/2944079-e7f2) SIP/2944079-e7f2 [EMAIL PROTECTED]:2 Up Dial(SIP/2944093|36|tr) 2 active channels 1 active call hestia*CLI hestia*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Form Hold Last Message xxx.yyy.128.115 (None) e77bba33-cc 00101/02261 unkn No Rx: REGISTER xxx.yyy.128.110 (None) 739f4603-e8 00101/00778 unkn No Rx: REGISTER xxx.yyy.128.86 (None) 56caad3a-eb 00101/01046 unkn No Rx: REGISTER xxx.yyy.128.115 (None) 91ea0410-60 00101/02262 unkn No Rx: REGISTER xxx.yyy.128.86 (None) 488801e-105 00101/01046 unkn No Rx: REGISTER xxx.yyy.128.86 (None) c3b27274-ef 00101/01194 unkn No Rx: REGISTER xxx.yyy.128.77 2944093 2405f1ef74d 00102/0 ulaw No Tx: ACK xxx.yyy.128.83 2944079 cf1722ef-cc 00101/2 ulaw No Rx: ACK Doug. ___ --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 ___ --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] asterisk load balance
According to Kevin Fleming, this is not supported. -Original Message- From: unplug [mailto:[EMAIL PROTECTED] Sent: Tue 6/20/2006 10:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] asterisk load balance I am confusing where the asterisk should store the register information in realtime mode. As in my configuration, UA1 asterisk1 + UA2 asterisk2 + database UA3 - asterisk3 + 3 UAs connected to 3 asterisk with a common database to store user information and dial plan. However, asterisk1 seems doesn't know there are UA2 and UA3 already registered in the system. I wonder the register information should be store in DB. When there is a invite request, asterisk will query the database and find out the calling party contact information. Am I right? But in the case above, asterisk only know the UA which register to it. Anyone can tell me the real mechanism of realtime for the UA registration? How and where asterisk to get the user registration when there is an invite comming? On 6/18/06, Aaron Daniel [EMAIL PROTECTED] wrote: On Sat, 17 Jun 2006, Douglas Garstang wrote: Good grief I hate Outlook webmail. I can't reply inline. Switch to thunderbird ;) Anyway, I disagree that all state info except hinting can be replicated. What about call transfers? If a call is sitting on pbx1, and the user transfers a call, if it goes to pbx2, Asterisk will complain that it cannot transfer the call as it doesn't know anything about it Well, I'm not sure what the problem with call transfers is. We have two registration servers, in which the phones can and do register with either server. If one phone makes a call on one server, they can complete the call with anyone else on their server, plus anyone on the other servers. The server just treats the transfer and bridge like any other phone call. If the phone is on another server, it hands off the conversation to that server after the transfer. And I think I'll address your NFS problems. Are you doing that for redundancy's sake or just for MWI? If it's just for MWI, then you might be better off setting up some scripts that drop some msg.txt files in the user's voicemail box on the registration servers. No need to replicate registration to the voicemail server, that's just extra unneeded traffic. Plus, with something like that, you don't have to worry about the voicemail nfs share dying and bringing down the asterisk network. If it's for redundancy, set up another voicemail server or two, and use DRBD or some sort of sync tool between them, with the MWI script and you'll have fixed the redundancy problem. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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 ___ --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] MySQL Realtime Voicemail Connection Lost
I'm using realtime for voicemail users, and for reasons that I don't yet understand, when it doesn't get used for a while (like overnight), the first connection attempt of the day will display this on the console. Jun 21 07:54:00 ERROR[8112]: cdr_addon_mysql.c:159 mysql_log: cdr_mysql: Unknown connection error: (2006) MySQL server has gone away Jun 21 07:54:01 NOTICE[8120]: rtp.c:564 ast_rtp_read: Unknown RTP codec 96 received -- Executing VoiceMail(SIP/xxx.187.142.186-b773c428, [EMAIL PROTECTED]) in new stack Jun 21 07:54:01 ERROR[8120]: res_config_mysql.c:623 mysql_reconnect: MySQL RealTime: Failed to reconnect. Check debug for more info. Jun 21 07:54:01 WARNING[8120]: app_voicemail.c:2411 leave_voicemail: No entry in voicemail config file for '2944017' The next connection attempt will work. Happens like clockwork every morning. It would seem that Asterisk is not reconnecting the first time, even when it says it is. I'm thinking I may open a bug on this. Has anyone else encountered this behaviour? Doug. ___ --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] Conferencing with multiple servers
-Original Message- From: Patrick [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 20, 2006 12:05 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Conferencing with multiple servers On Tue, 2006-06-20 at 15:22 +0100, Wildheart wrote: Hi, I am trying to join 2 asterisk servers together using a sip channel. This is so, if a user joins a conference on box A and another user joins a conference on box B, providing they are in the same conference room, the two conferences are joined via the sip channel. We only want to join the conferences together if they have users in them and we don't want to point all the conferences to one server as we would like to try to balance the load a bit. This is a general problem with the 'enterprise grade' aspects of Asterisk. As far as I know, there is no way to distribute applications (eg: Queue, Meetme etc) between multiple Asterisk systems. You really need to run the applications that will serve a common set of phones on the same Asterisk system, and then fail over to a secondary if necessary. Doug. ___ --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] Polycom Buddies in 1.6.6
All, Slightly off topic. Polycom released their SIP software version 1.6.6 for their phones recently. I was under the impression that this release fixed a previous limitation where the phones would only watch 7 buddies, ie send 7 sip subscriptions to Asterisk. I have configured a phone directory to watch 30 or so appearances, and it still seems to only be sending 7 subscriptions to Asterisk. Has anyone else got this to work? Doug. ___ --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] Voicemail with NFS
I have some experience with fibre-channel. I wouldn't be surprised if Asterisk behaved in exactly the same way if a fibre-channel volume went offline. It's also prohibitively expensive. -Original Message- From: Avi Miller [mailto:[EMAIL PROTECTED] Sent: Sat 6/17/2006 1:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Voicemail with NFS Douglas Garstang wrote: I don't think unison is a workable solution. It doesn't scale. The network and system load would increase exponentially as we added asterisk servers to our cluster. If you're clustering that many boxes, I'd investigate fibre channel SAN and GFS. That way, each node of the cluster just mounts the voicemail location locally. -- National Manager - Special Projects Melbourne / Sydney / Canberra / Hobart / London / 2/340 Gore StreetT: 1 300 SQUIZ (77859) Fitzroy, VIC T: 03 9486 0411 3065 F: 03 9486 0611 W: http://www.squiz.net/ . Open Source - Own it - Squiz.net ./ ___ --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
RE: [Asterisk-Users] Voicemail with NFS
Yes, we'd need it on every single box. We had a dedicated voicemail server in the first place. I decided to distribute voicemail between all boxes because the script that I had that copied the phone registrations over to the voicemail server (for mwi) was unreliable. -Original Message- From: Simon Woodhead [mailto:[EMAIL PROTECTED] Sent: Sat 6/17/2006 1:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Voicemail with NFS We use Unison Doug and it works just fine. It isn't perfect in theory but we've had no issues in practice. Your concerns over sacalbility are resolved by implementation - do you need it on every single Asterisk box, or maybe local to just two with routing to them and failover in the dial-plan? Unison is like two way rsync and consequently extremely efficient. Simon On 6/17/06, Douglas Garstang [EMAIL PROTECTED] wrote: Mike, I don't think unison is a workable solution. It doesn't scale. The network and system load would increase exponentially as we added asterisk servers to our cluster. Doug. -Original Message- From: Mike Diehl [mailto:[EMAIL PROTECTED] Sent: Fri 6/16/2006 9:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Voicemail with NFS I don't know how big your voicemail system is, but have you considered using Unison to syncronize the vm accross all your servers? I'm deploying multiple servers with two vm servers, each sync'ed every 5? minutes. If one fails, the other one should be good enough. Just a though, Mike On Friday 16 June 2006 16:14, Brian Capouch wrote: Douglas Garstang wrote: Douglas Garstang wrote: I hope someone isn't going to tell me that the voicemail directory going away is going to cause Asterisk to fall in a heap on the floor. Brian Capouch wrote: You never give up on dissing Asterisk, do you, Pococurante? This would be acceptable behaviour for you? An NFS-mounted volume isn't ever going to be as reliable as one mounted on the local filesystem. You are introducing additional points of failure both with respect to there now being two hard drives involved, as well as an interposed network that can fail in a variety of ways. So by definition this arrangement isn't going to be as reliable as one based on a native filesystem. And you never have answered the direct question: what do you expect the logical thing would be to happen if all the sudden an important system resource has just gone away? Regardless of the answer (because a rejoinder to that would then be, So add that behavior into Asterisk, or help the developers do so . . ) my point isn't that you are finding--actually looking for--places where catastrophic behavior makes Asterisk suffer. The problem is that you don't ever say, So what are some reasonable things that might be done in this situation; instead you emit a scathing remark (fall in a heap on the floor) that would indicate you've discovered some glaring design flaw that any idiot would have known to design around ahead of your finding it. It is not automatically the case that if Asterisk doesn't do something you think it should do it means that Asterisk is horribly and glaringly flawed. But that's what you *always* assume, and you always--ALWAYS--do so snidely
RE: [Asterisk-Users] Voicemail with NFS
JR, Are you sure that a ro mounted volume won't behave in the same fashion as a rw mounted one when the NFS server is abruptly shut down? Have you tried shutting down the NFS server? Does Asterisk recover from this? Doug. -Original Message- From: JR Richardson [mailto:[EMAIL PROTECTED] Sent: Sat 6/17/2006 8:04 AM To: asterisk-users@lists.digium.com Cc: Subject: Re: [Asterisk-Users] Voicemail with NFS Pococurante! Or Pococurante? Or you're a big fat poco! Damn Brian, I had to look this word up. SYLLABICATION: po·co·cu·ran·te Pronunciation: (pōkō-koo-ran'tē, -rän'-, -kyoo-; It.pôkô-kOO-rän'te) ADJECTIVE: Indifferent; apathetic; nonchalant. NOUN: One who does not care; a careless or indifferent person. Thanks for expanding my vocabulary. I actually export my NFS share from my Voicemail server read-only. The registration servers mount the VFS share, only to get the MWI function working. I send all VM functions over the VM server so the registration servers never have to write to the VM NSF share so Asterisk doesn't care if it drops off. Digium has been working on a remote MWI over IAX, hopefully it will be completed soon and there will be no need for NFS or SAMBA to do any of this. There are 2 patches on Mantis that do just this, two different implementations and the authors report great success. Mantis issue 4236 and 4371 JR Poco-not ___ --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
RE: [Asterisk-Users] asterisk load balance
Good grief I hate Outlook webmail. I can't reply inline. Anyway, I disagree that all state info except hinting can be replicated. What about call transfers? If a call is sitting on pbx1, and the user transfers a call, if it goes to pbx2, Asterisk will complain that it cannot transfer the call as it doesn't know anything about it -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Fri 6/16/2006 11:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [Asterisk-Users] asterisk load balance On Fri, 16 Jun 2006, Douglas Garstang wrote: Unless you can guarantee that the system that is currently processing a call will be the system that handles a transfer request from a phone, are the same, then transfers will not work. Incorrect. Transfers work fine between multiple asterisk boxes. Round robin DNS won't work at all. Every time you send out a SIP message, your going to be sending it to a different Asterisk box. For example, your initial INVITE will go to asterisk server 1. Asterisk server 1 will then send back a message requesting authorisation. Your phone does another lookup, and gets Asterisk server 2 this time. The phone sends the new INVITE with the auth info to Asterisk server 2. Asterisk server 2 will probably be ok with this, but when it sends a TRYING back to the phone, depending on the phone you are using, everything will fall in a heap on the floor. I know polycoms do. They get this TRYING from an asterisk server they didn't send and they go 'huh?'. This is entirely phone dependant, and usually the phones that fall in a heap (like the phrase much?) also handle secondary server configurations MUCH better than the phones that don't. Polycoms and sipura's handle SRV and backup server settings better than cisco's, but cisco's won't jump from server to server. I'm sure most other stuff will fail too. The Asterisk boxes share no state information. It's all in how you program the dial plan. The main thing that doesn't share state information that may cause problems is hinting. Everything else is programmable somewhere in the system :) -Original Message- From: unplug [mailto:[EMAIL PROTECTED] Sent: Fri 6/16/2006 9:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: [Asterisk-Users] asterisk load balance Hi, I am designing a asterisk load balancing model as follow. There are 3 asterisks connected to a single DB and a single server storing all the configuration file and voicemail. Round Robin DNS will distribute the request to asterisks. DNS round robin ---+ asterisk1--+ DB and file server +---asterisk2---+ +---asterisk3---+ Your design would work just fine as long as you have your dialplan is configured right. Keep in mind though that if asterisk1 dies, you just lost your db. Does anyone has load balancing experience implemented in asterisk that can share? Does my design work? Does any conflict will happen in my design? Any comment? Thanks! ___ --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 -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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
RE: [Asterisk-Users] Voicemail with NFS
Other applications can handle it. Don't see why Asterisk can't. Mount the nfs volume with the -soft option. Do a 'df -k' and you will see that the df command will time out in a couple of seconds. Why can't Asterisk do the same? -Original Message- From: Ira [mailto:[EMAIL PROTECTED] Sent: Sat 6/17/2006 3:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [Asterisk-Users] Voicemail with NFS At 03:44 PM 6/16/2006, you wrote: The hanging waiting for NFS volume to become avaiable is a classic NFS situation, hardly limited to your little experiment. Silly question, but how is this different than a hard disk in the local machine crashing or the router dying or even pulling the plug on the * box itself? Would you expect it to handle any of those scenarios? Ira ___ --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] Zaptel HZ Warning
Anyone else get this while compiling zaptel? I'm guessing I have to modify my kernel. Neato. :( Does that mean that the zaptel module (I'm really after ztdummy), or this xpp_zap thing won't be usable...? Not that I have zaptel hardware, but it seems Asterisk won't compile itself without zaptel being installed. CC [M] /root/software/zaptel-1.2.6/xpp/xpp_zap.o /root/software/zaptel-1.2.6/xpp/xpp_zap.c:365:2: warning: #warning This module will not be usable since the kernel HZ setting is not 1000 ticks per second. /root/software/zaptel-1.2.6/xpp/xpp_zap.c:365:2: warning: #warning This module will not be usable since the kernel HZ setting is not 1000 ticks per second. CC [M] /root/software/zaptel-1.2.6/xpp/zap_debug.o Doug. ___ --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] Executing a Function from AGI
-Original Message-
From: Time Bandit [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 15, 2006 4:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Executing a Function from AGI
I'm getting nowhere with this. Is it even possible to set a
variable to the result of a function call in AGI???
snip
SET VARIABLE DIALPATH ${DUNDILOOKUP(2944093|180net)}
in both cases, DIALPATH is set to a literal
${DUNDILOOKUP2944093|180net}
What am I doing wrong here?
You are telling it to assign the value ${DUNDILOOKUP2944093|180net} to
the variable DIALPATH, and it seems it is doing exactly that
Remember that you're in an AGI, not in the dialplan, so your variable
doesn't get interpreted
And to answer your question, I think you should call the function, get
the result, then assign that to your variable
Still not having any luck! I tried sending this to stdout:
SET VARIABLE DIALPATH DUNDILOOKUP(2944093|180net)
I must be missing something still. I assigned DIALPATH to the function.
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RE: [Asterisk-Users] Re: Executing a Function from AGI
-Original Message-
From: Stefan Tichy [mailto:[EMAIL PROTECTED]
Sent: Friday, June 16, 2006 7:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Re: Executing a Function from AGI
On Thu, Jun 15, 2006 at 03:21:32PM -0600, Douglas Garstang wrote:
I've tried this:
EXEC Set DIALPATH=${DUNDILOOKUP(2944093|180net)}
and also:
SET VARIABLE DIALPATH ${DUNDILOOKUP(2944093|180net)}
in both cases, DIALPATH is set to a literal
${DUNDILOOKUP2944093|180net}
get full variable evaluates a channel expression, but set
variable cannot be used this way.
Use GET FULL VARIABLE to get the value and then use SET VARIABLE
to store this value in the DIALPATH variable.
Oh... Thanks... This doesn't seem to be documented anywhere.
Where did you find out about this?
Doug.
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[Asterisk-Users] Voicemail with NFS
I have /var/spool/asterisk/voicemail NFS mounted from another server. Everything is fine, until I simulate an NFS server failure, by shutting down the NFS server process. At this point, Asterisk becomes almost non-responsive. It won't even process a 'sip show peers' command correctly. It displays a few lines of text, pauses for several seconds, and then displays the rest. When a call comes into the system, Asterisk seems to do nothing for several seconds, and generally acts really sluggish. The phone gives up after several seconds, because Asterisk isn't doing anything. I have used the soft option with the NFS mount. I hope someone isn't going to tell me that the voicemail directory going away is going to cause Asterisk to fall in a heap on the floor. We just changed our model from a single, central voicemail server, to a distributed model, to get around some issues. We can't lose ALL pbx functionality just because the voicemail NFS server goes away. That's insane. Doug. ___ --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] Voicemail with NFS
I'll give this a try, but what happens when someone tries to access their voicemail? Common sense would say that THEN the system will fall apart, which isn't much of a solution. Doug. -Original Message- From: Bruce Ferrell [mailto:[EMAIL PROTECTED] Sent: Friday, June 16, 2006 2:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voicemail with NFS you might want to try autofs to drive the nfs functions. it'll make you less susceptable as the filesystem won't be mounted full time Douglas Garstang wrote: I have /var/spool/asterisk/voicemail NFS mounted from another server. Everything is fine, until I simulate an NFS server failure, by shutting down the NFS server process. At this point, Asterisk becomes almost non-responsive. It won't even process a 'sip show peers' command correctly. It displays a few lines of text, pauses for several seconds, and then displays the rest. When a call comes into the system, Asterisk seems to do nothing for several seconds, and generally acts really sluggish. The phone gives up after several seconds, because Asterisk isn't doing anything. I have used the soft option with the NFS mount. I hope someone isn't going to tell me that the voicemail directory going away is going to cause Asterisk to fall in a heap on the floor. We just changed our model from a single, central voicemail server, to a distributed model, to get around some issues. We can't lose ALL pbx functionality just because the voicemail NFS server goes away. That's insane. Doug. ___ --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 -- One day at a time, one second if that's what it takes ___ --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
RE: [Asterisk-Users] Voicemail with NFS
-Original Message- From: Brian Capouch [mailto:[EMAIL PROTECTED] Sent: Friday, June 16, 2006 3:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voicemail with NFS Douglas Garstang wrote: I'll give this a try, but what happens when someone tries to access their voicemail? Common sense would say that THEN the system will fall apart, which isn't much of a solution. What do you want it to do? The hanging waiting for NFS volume to become avaiable is a classic NFS situation, hardly limited to your little experiment. There are NFS mount options that stop the hang, but they won't correct the fact that your voicemail store isn't there. What do I want what to do? I don't see why Asterisk should hang. As I said, I mounted the NFS share with the 'soft' option and a timeout of 5 seconds. When I do a 'df -k' command, the result hangs for 5s or so and then returns. Asterisk doesn't seem to behave in the same way. Considering it's a function of the mount command, you'd think it would be transparent to Asterisk. Yes... soft causes it not to hang for everything else except Asterisk I don't care if the voicemail store goes away. I do care if the voicemail store going away causes all calls across the entire system to fail. Doug. ___ --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] Voicemail with NFS
-Original Message- From: Brian Capouch [mailto:[EMAIL PROTECTED] Sent: Friday, June 16, 2006 3:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voicemail with NFS Douglas Garstang wrote: I hope someone isn't going to tell me that the voicemail directory going away is going to cause Asterisk to fall in a heap on the floor. You never give up on dissing Asterisk, do you, Pococurante? This would be acceptable behaviour for you? ___ --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] Voicemail with NFS
CF... I tried setting it to 60s, and it delays the same final result. The system becomes unresponsive, but it just takes a few more seconds to do it. Looks like it's somehow related to Asterisk polling it's voicemail store every checkmwi seconds. Well... :( -Original Message- From: C F [mailto:[EMAIL PROTECTED] Sent: Friday, June 16, 2006 4:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voicemail with NFS I'm curious is this might help: this is in sip.conf ;checkmwi=10; Default time between mailbox checks for peers This might be the reason asterisk misbehaves when the NFS mount is unavailable. Therfore I think this might tell asterisk not to try looking thru the folders every 10 seconds and will therefore allow asterisk to allow for a missing NFS mount for longer than 10 seconds. However if someone wants to leave a message, I'm not sure this will work. On 6/16/06, Douglas Garstang [EMAIL PROTECTED] wrote: -Original Message- From: Brian Capouch [mailto:[EMAIL PROTECTED] Sent: Friday, June 16, 2006 3:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voicemail with NFS Douglas Garstang wrote: I hope someone isn't going to tell me that the voicemail directory going away is going to cause Asterisk to fall in a heap on the floor. You never give up on dissing Asterisk, do you, Pococurante? This would be acceptable behaviour for you? ___ --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 ___ --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] Voicemail with NFS
-Original Message- From: Brian Capouch [mailto:[EMAIL PROTECTED] Sent: Friday, June 16, 2006 3:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voicemail with NFS Douglas Garstang wrote: I'll give this a try, but what happens when someone tries to access their voicemail? Common sense would say that THEN the system will fall apart, which isn't much of a solution. What do you want it to do? The hanging waiting for NFS volume to become avaiable is a classic NFS situation, hardly limited to your little experiment. There are NFS mount options that stop the hang, but they won't correct the fact that your voicemail store isn't there. We've just tried the same thing, but with a samba server. Shut down the samba server, and Asterisk stops processing calls. Actually, this time the phones will fail over to the next asterisk box. However, if all asterisk boxes are pointing to the same samba server, then failover from one asterisk box to another isn't going to help at all. So... Asterisk responds to a samba failure in the same way as an NFS failure. Looks like it's time to start playing with rsync... ___ --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] Voicemail with NFS
Mike, Never heard of Unison... do you have a link to it? Doug. -Original Message- From: Mike Diehl [mailto:[EMAIL PROTECTED] Sent: Friday, June 16, 2006 9:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voicemail with NFS I don't know how big your voicemail system is, but have you considered using Unison to syncronize the vm accross all your servers? I'm deploying multiple servers with two vm servers, each sync'ed every 5? minutes. If one fails, the other one should be good enough. Just a though, Mike On Friday 16 June 2006 16:14, Brian Capouch wrote: Douglas Garstang wrote: Douglas Garstang wrote: I hope someone isn't going to tell me that the voicemail directory going away is going to cause Asterisk to fall in a heap on the floor. Brian Capouch wrote: You never give up on dissing Asterisk, do you, Pococurante? This would be acceptable behaviour for you? An NFS-mounted volume isn't ever going to be as reliable as one mounted on the local filesystem. You are introducing additional points of failure both with respect to there now being two hard drives involved, as well as an interposed network that can fail in a variety of ways. So by definition this arrangement isn't going to be as reliable as one based on a native filesystem. And you never have answered the direct question: what do you expect the logical thing would be to happen if all the sudden an important system resource has just gone away? Regardless of the answer (because a rejoinder to that would then be, So add that behavior into Asterisk, or help the developers do so . . ) my point isn't that you are finding--actually looking for--places where catastrophic behavior makes Asterisk suffer. The problem is that you don't ever say, So what are some reasonable things that might be done in this situation; instead you emit a scathing remark (fall in a heap on the floor) that would indicate you've discovered some glaring design flaw that any idiot would have known to design around ahead of your finding it. It is not automatically the case that if Asterisk doesn't do something you think it should do it means that Asterisk is horribly and glaringly flawed. But that's what you *always* assume, and you always--ALWAYS--do so snidely. Pococurante. B. ___ --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
RE: [Asterisk-Users] Voicemail with NFS
Mike, I don't think unison is a workable solution. It doesn't scale. The network and system load would increase exponentially as we added asterisk servers to our cluster. Doug. -Original Message- From: Mike Diehl [mailto:[EMAIL PROTECTED] Sent: Fri 6/16/2006 9:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Voicemail with NFS I don't know how big your voicemail system is, but have you considered using Unison to syncronize the vm accross all your servers? I'm deploying multiple servers with two vm servers, each sync'ed every 5? minutes. If one fails, the other one should be good enough. Just a though, Mike On Friday 16 June 2006 16:14, Brian Capouch wrote: Douglas Garstang wrote: Douglas Garstang wrote: I hope someone isn't going to tell me that the voicemail directory going away is going to cause Asterisk to fall in a heap on the floor. Brian Capouch wrote: You never give up on dissing Asterisk, do you, Pococurante? This would be acceptable behaviour for you? An NFS-mounted volume isn't ever going to be as reliable as one mounted on the local filesystem. You are introducing additional points of failure both with respect to there now being two hard drives involved, as well as an interposed network that can fail in a variety of ways. So by definition this arrangement isn't going to be as reliable as one based on a native filesystem. And you never have answered the direct question: what do you expect the logical thing would be to happen if all the sudden an important system resource has just gone away? Regardless of the answer (because a rejoinder to that would then be, So add that behavior into Asterisk, or help the developers do so . . ) my point isn't that you are finding--actually looking for--places where catastrophic behavior makes Asterisk suffer. The problem is that you don't ever say, So what are some reasonable things that might be done in this situation; instead you emit a scathing remark (fall in a heap on the floor) that would indicate you've discovered some glaring design flaw that any idiot would have known to design around ahead of your finding it. It is not automatically the case that if Asterisk doesn't do something you think it should do it means that Asterisk is horribly and glaringly flawed. But that's what you *always* assume, and you always--ALWAYS--do so snidely. Pococurante. B. ___ --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
RE: [Asterisk-Users] asterisk load balance
Unless you can guarantee that the system that is currently processing a call will be the system that handles a transfer request from a phone, are the same, then transfers will not work. Round robin DNS won't work at all. Every time you send out a SIP message, your going to be sending it to a different Asterisk box. For example, your initial INVITE will go to asterisk server 1. Asterisk server 1 will then send back a message requesting authorisation. Your phone does another lookup, and gets Asterisk server 2 this time. The phone sends the new INVITE with the auth info to Asterisk server 2. Asterisk server 2 will probably be ok with this, but when it sends a TRYING back to the phone, depending on the phone you are using, everything will fall in a heap on the floor. I know polycoms do. They get this TRYING from an asterisk server they didn't send and they go 'huh?'. I'm sure most other stuff will fail too. The Asterisk boxes share no state information. Doug. -Original Message- From: unplug [mailto:[EMAIL PROTECTED] Sent: Fri 6/16/2006 9:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: [Asterisk-Users] asterisk load balance Hi, I am designing a asterisk load balancing model as follow. There are 3 asterisks connected to a single DB and a single server storing all the configuration file and voicemail. Round Robin DNS will distribute the request to asterisks. DNS round robin ---+ asterisk1--+ DB and file server +---asterisk2---+ +---asterisk3---+ Does anyone has load balancing experience implemented in asterisk that can share? Does my design work? Does any conflict will happen in my design? Any comment? Thanks! ___ --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
RE: [Asterisk-Users] DUNDi Not Able to Handle Complex FailoverSituations
-Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Wednesday, June 14, 2006 7:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DUNDi Not Able to Handle Complex FailoverSituations On Wed, 14 Jun 2006, Douglas Garstang wrote: Why doesn't the DUNDILOOKUP function return the weight of a path to a number? The CLI 'dundi lookup' command does. What about the mac address and expiry period? The CLI command returns those, but the DUNDILOOKUP function does not. Why? Correct me if I'm wrong, but DUNDi is doing all the failover work for you. It decides based on the weights what route is best. If you want one route to be higher than another, set it up that way. That's the benefit of using DUNDILOOKUP to handle it, no more work for you after the initial routing. DUNDi does not handle the situation of phone failover as well as static numbers (ie queues), which is what we are trying to acheive. If that doesn't work for you, program the routes directly into the dialplan instead of using DUNDi, it seems like you'll get better results that way. We did that for a while until we decided to move to DUNDi. Some people will find it more suited to their needs, some won't. There are no routes. Termination ends at the Asterisk box. ___ --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] DUNDi Not Able to Handle Complex FailoverSituations
-Original Message- From: Watkins, Bradley [mailto:[EMAIL PROTECTED] Behalf Of Watkins, Bradley Sent: Thursday, June 15, 2006 2:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] DUNDi Not Able to Handle Complex FailoverSituations Unless I'm misunderstanding what you're looking to do, Aaron has hit the nail on the head here. You need to set it up so that the secondary, tertiary, etc. boxes are weighted differently. That way, you need not know or care about the weights directly within the dialplan. It isn't as simple as that. When a failure occurs, we only want to use a DUNDi route when it's the primary for a queue. ___ --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] Asterisk Realtime and SIP Registration
Kevin Fleming has said on numerous ocassions that this is known not to work, and is not supported. -Original Message-From: Benjamin Stocker [mailto:[EMAIL PROTECTED]Sent: Tuesday, June 06, 2006 4:31 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk Realtime and SIP RegistrationHi!I use the following configuration to register my asterisk server to my SIP provider:register = 12345:[EMAIL PROTECTED]/12345sip.conf :[sipout-test]type=peerusername=12345fromuser=12345fromdomain=provider.comsecret=passwdinsecure=veryhost=sip.provider.com qualify=yescontext=test-incomingextensions.conf:exten = 12345,1,Dial(SIP/10)exten = _0NXZXX,1,Dial(SIP/[EMAIL PROTECTED])This works fine when I put it into the config files. I can dial other numbers via my provider and receive calls. Wenn I put everything into Realtime tables (except the register command), incoming calls work only after * I make at least one outgoing call - or - * Somebody calls me twiceOn incoming calls, the caller first gets a 'user unavailale' from my SIP provider. When hanging up and calling again, the connection establishes successfully and I see this when entering 'sip show peers': sipout-test/12345 IP.AD.DR.ESS 5060 UNKNOWNThis line does not show up when I registering my phone to my asterisk server. But it shows up immediately after registerung the phone when I use config files instead of RTA. I don't know wheter this is RTA- or a config-problem. ___ --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] Distributed ACD Queues
It
seems that I am having a heck of a time explaining my attempts at distributing
ACD Queues to the list. Here's one little problem, that's only a piece of the
puzzle.
dundi.conf:
180q
=
global_dundi_q_pbx1,100,IAX,dundi1:[EMAIL PROTECTED]/${NUMBER},nopartial180q
=
global_dundi_q_pbx2,200,IAX,dundi2:[EMAIL PROTECTED]/${NUMBER},nopartial180q
=
global_dundi_q_pbx3,300,IAX,dundi3:[EMAIL PROTECTED]/${NUMBER},nopartial
extensions.conf(PBX1):
[global_dundi_q_pbx1]include = one_queue_acd
[global_dundi_q_pbx2][global_dundi_q_pbx3]
extensions.conf(PBX2):
[global_dundi_q_pbx1]
[global_dundi_q_pbx2]
include = one_queue_acd[global_dundi_q_pbx3]
extensions.conf(PBX3):
[global_dundi_q_pbx1]
[global_dundi_q_pbx2][global_dundi_q_pbx3]
include = one_queue_acd
[one_queue_acd]
exten
= 2944000,1,
Our
polycom phones are registering to a primary Asterisk system. It's ESSENTIAL that
queue calls for a given company go to the SAME box as the phones are registered
to. Queues won't work correctly if they're are split between
servers.
If a
phone registered to pbx1 wants to call the queue at 2944000, the call comes into
pbx1. If I do a dundi lookup on that number on the console I get
this:
hestia*CLI dundi lookup 2944000@180q 1. 200
IAX2/dundi2:[EMAIL PROTECTED]/oe_main
(EXISTS) from 00:0e:0c:a1:90:82, expires in 0
s 2. 300
IAX2/dundi3:[EMAIL PROTECTED]/oe_main
(EXISTS) from 00:0e:0c:a1:92:6f, expires in 0
sDUNDi lookup completed in 63 ms
If I
do the dundi lookup in the dialplan, all I get is
"IAX2/dundi2:[EMAIL PROTECTED]/oe_main" with no weight.
DUNDi never returns local matches (with a weight of 100 in this case).
whichis a problem.The result I get from the DUNDi lookup in the
dialplan is astring that points to the SECONDARY server. I don't want to
send the call to the secondary server!!!
I
could first do a local lookup with ChanIsAvail and look in the correct context
(global_dundi_q_pbx1 etc) to see if the number is local first. However, this is
tricky as we're trying to maintain a common dialplan between all our servers.
I'd therefore have to probe the system by executing an external command and
pulling the hostname or something just so I know which context to look
in.
I'm
guessing no one has tried to distribute acd queues before...
___
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[Asterisk-Users] ACD Distributed Scenario....
We need to make sure that all queue applications run on the correct system that the user agents that own the queue application are registered to. So when a server fails and the user agents register with their secondary server (which will always be configured to be the same server for those related agents) the queue application is running on that server and routed to correctly by it's peers. Enters DUNDi: Working scenario: 1) Configured 3 contexts, referenced by DUNDi, to manage which server is the primary, secondary, and tertiary server for each given queue. So: a. UA1, 2, and 3 register with Astbox1 as their primary server b. Their registration tables refer to Astbox2 as their secondary registration server and Astbox3 as their tertiary registration server c. Agents are logging into the queue1 via UA1, 2, and 3 d. Queue1's dial plan logic is in the same context on all boxes e. Queue1's dial plan logic is referred to via 3 different DUNDi contexts weighted according to which server is the primary, secondary, and tertiary host server for the user agents (UA1,2, and 3) f.So queue1, assigned the phone number of 5551212, is assigned to the Primary DUNDi context on Astbox1 with the weight of 0 g. Then queue1 is assigned to the secondary DUNDi context on Astbox2 with the weight of 100 and to the tertiary DUNDi context on Astbox3 with the weight of 200 h. So let's say we make a call from an User Agent on Astbox2 to 5551212 i. When determining which server to terminate a call to 5551212 on we do a local lookup first on Astbox2 to see if the primary server for that number happens to be the server performing the routing logic... if so, we directly route the call to that queue on the local server j.In this case Astbox2 does not refer to queue1 in the primary DUNDi context, Astbox1 refers to queue1 in it's primary context, so we do a DUNDi lookup to find the next server we should route the call to k. Due to weighting, we receive the IP of Astbox1 as the first DUNDi destination and the IP of Astbox3 as a second DUNDi destination serving that queue and we route the call to the first destination IP l. Everything is fine... but when the primary server fails (Astbox1) and the the secondary server happens to be the box that is routing the call (Astbox2) there is a logic gap we need help addressing 2) Logic gap we need to address a. UA1, 2, and 3 normally register with Astbox1 as their primary server but it has now failed b. So UA1, 2, and 3 now register with Astbox2 c. Due to queue1's routing logic, that the agents assigned to UA1, 2, and 3 log into, residing in the same context on all boxes we are able to handle calls to that context on Astbox2 (please refer to our statement in 1.d through 1.g to re-paraphrase the queue and agent relationship) d. So let's say we make a call from a user agent Astbox2 to 5551212 e. When determining which server to terminate a call to 5551212 on we do a local lookup first on Astbox2 to see if the primary context shows that number as local (queue1)... if so, we directly route the call to that queue on the local server f.In this case it is not because the context we are referring to (PRIMARY) does not reside on Astbox2 with 5551212 in it's context so we do a DUNDi lookup to find the next server we should route the call to g. Due to weighting and the fact that Astbox1 has failed, we receive the IP of Astbox3 as the only DUNDi destination serving that queue (HERE IS THE PROBLEM --- Astbox3 is tertiary... the box during this failed condition that has the highest weight is the box doing the lookups (Astbox2) along with Astbox2 is the box that the destination agents of the queue are now registered with) h. If we could actually query to find the weight of the DUNDi returned lookup we could add logic to the scripting to determine if it is the Primary server with a weight of 0. i. If not we could add some logic to see if the server we are on happens to be the secondary j.If so the call would then remain on that box k. If not we could then route to the returned IP address from the DUNDi lookup ___ --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] DUNDi Not Able to Handle ComplexFailoverSituations
-Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Thursday, June 15, 2006 9:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] DUNDi Not Able to Handle ComplexFailoverSituations On Thu, 15 Jun 2006, Douglas Garstang wrote: It isn't as simple as that. When a failure occurs, we only want to use a DUNDi route when it's the primary for a queue. Then don't use DUNDi for queues, use it just for the phones. Seriously, you obviously know exactly which servers you want to be primary for a certain queue, program it into the dialplan. DUNDi should only be used for DYNAMIC extensions, i.e. phones that may or may not be registered at the time of the call, phones that move, phones that register with different servers at different times. If you're deadset on using DUNDi for it, set up different DUNDi contexts so that you can say these queues are available here and these queues are available there. Honestly, it seems like a waste of server time to use DUNDi for something that you know is going to be on a particular server regardless of what happens. If we don't use DUNDi, then how are we going to get the Queue() application to follow the pbx server, and execute on the same Asterisk box that the phones are registered on? Doug. ___ --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] DUNDi Not Able to Handle ComplexFailoverSituations
-Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Thursday, June 15, 2006 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] DUNDi Not Able to Handle ComplexFailoverSituations On Thu, 15 Jun 2006, Douglas Garstang wrote: DUNDi does not handle the situation of phone failover as well as static numbers (ie queues), which is what we are trying to acheive. I'm confused, explain the phone failover not working to me. We need our queue application to follow the primary pbx server for a set of phones within a company. See my 'ACD Distributed Scenario' post made a little earlier for a full explanation. Doug. ___ --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] DUNDi Not Able to HandleComplexFailoverSituations
-Original Message- From: Watkins, Bradley [mailto:[EMAIL PROTECTED] Sent: Thursday, June 15, 2006 10:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] DUNDi Not Able to HandleComplexFailoverSituations Is it possible for you to explain in more detail the situation involved. I'm still thinking that what you're trying to achieve can be done at least with the help of DUNDi weights, but I still don't think I have a full grasp of the solution you're crafting. Bradley, See my post titled 'ACD Distributed Scenario' made an hour or two ago. Here it is again, reposted. We need to make sure that all queue applications run on the correct system that the user agents that own the queue application are registered to. So when a server fails and the user agents register with their secondary server (which will always be configured to be the same server for those related agents) the queue application is running on that server and routed to correctly by it's peers. Enters DUNDi: Working scenario: 1) Configured 3 contexts, referenced by DUNDi, to manage which server is the primary, secondary, and tertiary server for each given queue. So: a. UA1, 2, and 3 register with Astbox1 as their primary server b. Their registration tables refer to Astbox2 as their secondary registration server and Astbox3 as their tertiary registration server c. Agents are logging into the queue1 via UA1, 2, and 3 d. Queue1's dial plan logic is in the same context on all boxes e. Queue1's dial plan logic is referred to via 3 different DUNDi contexts weighted according to which server is the primary, secondary, and tertiary host server for the user agents (UA1,2, and 3) f.So queue1, assigned the phone number of 5551212, is assigned to the Primary DUNDi context on Astbox1 with the weight of 0 g. Then queue1 is assigned to the secondary DUNDi context on Astbox2 with the weight of 100 and to the tertiary DUNDi context on Astbox3 with the weight of 200 h. So let's say we make a call from an User Agent on Astbox2 to 5551212 i. When determining which server to terminate a call to 5551212 on we do a local lookup first on Astbox2 to see if the primary server for that number happens to be the server performing the routing logic... if so, we directly route the call to that queue on the local server j.In this case Astbox2 does not refer to queue1 in the primary DUNDi context, Astbox1 refers to queue1 in it's primary context, so we do a DUNDi lookup to find the next server we should route the call to k. Due to weighting, we receive the IP of Astbox1 as the first DUNDi destination and the IP of Astbox3 as a second DUNDi destination serving that queue and we route the call to the first destination IP l. Everything is fine... but when the primary server fails (Astbox1) and the the secondary server happens to be the box that is routing the call (Astbox2) there is a logic gap we need help addressing 2) Logic gap we need to address a. UA1, 2, and 3 normally register with Astbox1 as their primary server but it has now failed b. So UA1, 2, and 3 now register with Astbox2 c. Due to queue1's routing logic, that the agents assigned to UA1, 2, and 3 log into, residing in the same context on all boxes we are able to handle calls to that context on Astbox2 (please refer to our statement in 1.d through 1.g to re-paraphrase the queue and agent relationship) d. So let's say we make a call from a user agent Astbox2 to 5551212 ___ --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] DUNDi Not Able to Handle ComplexFailoverSituations
-Original Message- From: Stephen Davies [mailto:[EMAIL PROTECTED] Sent: Thursday, June 15, 2006 11:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DUNDi Not Able to Handle ComplexFailoverSituations On 15/06/06, Douglas Garstang [EMAIL PROTECTED] wrote: Who said I was a C programmer? Speaking for myself, I just assumed that you understood that the behaviour of an open-source application was the result of contributed code. Your message read to me like something of a demand that someone fixed it. You are probably trying to do something pretty fancy in your dialplan and that probably brings requirements that the original authors didn't foresee. They are scratching their itch. As you said, DUNDi was Mark's initiative to make a open access call routing system, rather than to do with failover. If you can hack Asterisk dialplan code, then I think if you open that file, take a look at other code that sets variables (search for a variable name you know is set, like DIALSTATUS), do some cut and paste and you'll discover that, guess what: you ARE a C programmer. Actually, I'd say I'm not a C programmer. In Asterisk 1.2.7.1, in pbx_dundi.c, function dundi_lookup_exec(), I Added this line: pbx_builtin_setvar_helper(chan, DUNDWEIGHT, dr[x].weight); right below the two other lines that set the DUNDTECH and DUNDDEST variables. When I execute my DundiLookup application in the dialplan, the Asterisk console bombs out. I assume it's core dumping or something. I don't know why though as I only added another line like the ones above. The DUNDTECH and DUNDDEST variables are not being referenced anywhere else in any file. ALSO... The DundiLookup application command has been deprecated: Jun 15 12:44:14 WARNING[2935]: pbx_dundi.c:3872 dundi_lookup_exec: This application has been deprecated in favor of the DUNDILOOKUP dialplan function. In favour of the DUNDILookup function. The DUNDILookup function does NOT seem to set the DUNDTECH and DUNDDEST variables, so it seems we have in effect gone backwards in functionality. In any case, I guess I'll have to try and figure out how to modify the string that DUNDILookup returns, which I'm sure will be harder than adding a new variable to DundiLookup() Doug. ___ --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] DUNDILOOKUP and DundiLookup()
The DundiLookup() application command seems to have been replaced by the DUNDILOOKUP application function. I'm wondering why, because the DUNDILOOKUP function doesn't set the TECH and DEST variables. I edited the code and added a WEIGHT variable to the variables set, but the DUNDILOOKUP function doesn't seem to export _any_ of these. The DundiLookup() application returns deprecated errors when you try and use it. I couldn't find where to edit the dial string returned by the DUNDILOOKUP function. Even if I could, it'd be messy, because then I'd have to extract the weight from the string before I dialled it. Getting what I need from variables is really a much better way. Doug. ___ --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] DUNDi Not Able to Handle ComplexFailoverSituations
-Original Message- From: Douglas Garstang Sent: Thursday, June 15, 2006 12:51 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] DUNDi Not Able to Handle ComplexFailoverSituations -Original Message- From: Stephen Davies [mailto:[EMAIL PROTECTED] Sent: Thursday, June 15, 2006 11:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DUNDi Not Able to Handle ComplexFailoverSituations On 15/06/06, Douglas Garstang [EMAIL PROTECTED] wrote: Who said I was a C programmer? Speaking for myself, I just assumed that you understood that the behaviour of an open-source application was the result of contributed code. Your message read to me like something of a demand that someone fixed it. You are probably trying to do something pretty I get annoyed Stephen when Digium goes around calling Asterisk 'enterprise grade', which in my opinion it really isn't. I'd consider distributed ACD queues to be a requirement for an enterprise grade product, but it's becoming apparent that there is no mechanism for implementing this. I'm being told that DUNDi isn't the right man for the job. ___ --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] DUNDi Not Able to HandleComplexFailoverSituations
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 15, 2006 12:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] DUNDi Not Able to
HandleComplexFailoverSituations
On Thu, 15 Jun 2006, Douglas Garstang wrote:
We need our queue application to follow the primary pbx
server for a set of phones within a company. See my 'ACD
Distributed Scenario' post made a little earlier for a full
explanation.
OK, let me get this straight.
You want the phones on the SAME server to hit the queues on
THAT server
only. Right?
Unless there's a server failure.
If that's right, then why use DUNDi for the queues, just set up an
extension (i.e. the queue entry point) that goes straight
into the queue
instead of using DUNDi for it, which adds more logic to
something VERY
simple. Since the phones are registered to that server,
obviously they
will drop into the local queue and not some random one.
Have a read of the post 'Distrubuted ACD Scenario' posted earlier. It really
explains it clearly, and states what the sticking point is. Also have a read of
Bradley Watkins post. He seems to have a grasp of it, and doesn't see a simple
solution.
You're making something dynamic that really shouldn't be
dynamic at all.
When the failover happens, the new primary server will have
the queue set
up, and anyone calling in will be calling into the queue on
that server.
Not necessarily. They might be calling in from a different server. We have to
ensure that we lookup the correct combination of primary/secondary server for
the queue, and what's actually available, and IAX the call over to THAT box to
process the Queue() command.
Now, if you're calling in from another server, i.e. someone outside
calling in, you can then use DUNDi with weights to drop them onto the
right server, but that's another story.
Finally, in order for the LOCAL server's DUNDi response to
show up, you
have to add the server to dundi.conf. So, so pbx1 has to be
in pbx1's
file, just like the other servers do.
No... this last bit doesnt. My dundi.conf has:
180q = global_dundi_q_pbx1,100,IAX,dundi1:[EMAIL PROTECTED]/${NUMBER},nopartial
180q = global_dundi_q_pbx2,200,IAX,dundi2:[EMAIL PROTECTED]/${NUMBER},nopartial
180q = global_dundi_q_pbx3,300,IAX,dundi3:[EMAIL PROTECTED]/${NUMBER},nopartial
What are you suggesting I change it to? Something like this?
180q = global_dundi_q_pbx1,100,IAX,dundi1:[EMAIL PROTECTED]/${NUMBER},nopartial
180q = global_dundi_q_pbx2,200,IAX,dundi2:[EMAIL PROTECTED]/${NUMBER},nopartial
180q = global_dundi_q_pbx3,300,IAX,dundi3:[EMAIL PROTECTED]/${NUMBER},nopartial
I really don't follow.
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RE: [Asterisk-Users] DUNDi Not Able to HandleComplexFailoverSituations
-Original Message-
From: Douglas Garstang
Sent: Thursday, June 15, 2006 1:23 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] DUNDi Not Able to
HandleComplexFailoverSituations
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 15, 2006 12:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] DUNDi Not Able to
HandleComplexFailoverSituations
On Thu, 15 Jun 2006, Douglas Garstang wrote:
We need our queue application to follow the primary pbx
server for a set of phones within a company. See my 'ACD
Distributed Scenario' post made a little earlier for a full
explanation.
OK, let me get this straight.
You want the phones on the SAME server to hit the queues on
THAT server
only. Right?
Unless there's a server failure.
If that's right, then why use DUNDi for the queues, just set up an
extension (i.e. the queue entry point) that goes straight
into the queue
instead of using DUNDi for it, which adds more logic to
something VERY
simple. Since the phones are registered to that server,
obviously they
will drop into the local queue and not some random one.
Have a read of the post 'Distrubuted ACD Scenario' posted
earlier. It really explains it clearly, and states what the
sticking point is. Also have a read of Bradley Watkins post.
He seems to have a grasp of it, and doesn't see a simple solution.
You're making something dynamic that really shouldn't be
dynamic at all.
When the failover happens, the new primary server will have
the queue set
up, and anyone calling in will be calling into the queue on
that server.
Not necessarily. They might be calling in from a different
server. We have to ensure that we lookup the correct
combination of primary/secondary server for the queue, and
what's actually available, and IAX the call over to THAT box
to process the Queue() command.
Now, if you're calling in from another server, i.e. someone outside
calling in, you can then use DUNDi with weights to drop
them onto the
right server, but that's another story.
Finally, in order for the LOCAL server's DUNDi response to
show up, you
have to add the server to dundi.conf. So, so pbx1 has to be
in pbx1's
file, just like the other servers do.
No... this last bit doesnt. My dundi.conf has:
180q =
global_dundi_q_pbx1,100,IAX,dundi1:[EMAIL PROTECTED]/${NUMBE
R},nopartial
180q =
global_dundi_q_pbx2,200,IAX,dundi2:[EMAIL PROTECTED]/${NUMBE
R},nopartial
180q =
global_dundi_q_pbx3,300,IAX,dundi3:[EMAIL PROTECTED]/${NUMBE
R},nopartial
What are you suggesting I change it to? Something like this?
180q =
global_dundi_q_pbx1,100,IAX,dundi1:[EMAIL PROTECTED]/
${NUMBER},nopartial
180q =
global_dundi_q_pbx2,200,IAX,dundi2:[EMAIL PROTECTED]/${NUMBE
R},nopartial
180q =
global_dundi_q_pbx3,300,IAX,dundi3:[EMAIL PROTECTED]/${NUMBE
R},nopartial
I really don't follow.
Ahh this reminds me too. If I am going to be getting the local system first
always, then I need to be able to return ALL the Dundi paths with the
DUNDILOOKUP function. It only returns one. How can I get DUNDILookup to return
every single path? It'd be great if they could return the weights for each too,
and then I could do my own logic.
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RE: [Asterisk-Users] DUNDi Not Able to HandleComplexFailoverSituations
Thanks Aaron. I got the local lookup to work. MIGHT have fixed our problem.
I ain't gonna poo my pants with excitement yet tho...
-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 15, 2006 1:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] DUNDi Not Able to
HandleComplexFailoverSituations
On Thu, 15 Jun 2006, Douglas Garstang wrote:
No... this last bit doesnt. My dundi.conf has:
180q =
global_dundi_q_pbx1,100,IAX,dundi1:[EMAIL PROTECTED]/${NUMBE
R},nopartial
180q =
global_dundi_q_pbx2,200,IAX,dundi2:[EMAIL PROTECTED]/${NUMBE
R},nopartial
180q =
global_dundi_q_pbx3,300,IAX,dundi3:[EMAIL PROTECTED]/${NUMBE
R},nopartial
What are you suggesting I change it to? Something like this?
180q =
global_dundi_q_pbx1,100,IAX,dundi1:[EMAIL PROTECTED]/
${NUMBER},nopartial
180q =
global_dundi_q_pbx2,200,IAX,dundi2:[EMAIL PROTECTED]/${NUMBE
R},nopartial
180q =
global_dundi_q_pbx3,300,IAX,dundi3:[EMAIL PROTECTED]/${NUMBE
R},nopartial
I really don't follow.
Here's an example. We have two primary call servers, both
are capable of
handling the call volume if one fails out. They're scm1 and scm2.
scm1 has a peer section for itself, so it shows up during
lookups. scm2
has a peer section for itself as well. They also have peer
sections for
each other and for the gateways:
scm1:
[00:E0:81:25:28:D3]
model = symmetric
host = sgw1
inkey = sgw1
outkey = scm1
include = all
permit = all
qualify = yes
[00:14:22:13:90:8D]
model = symmetric
host = scm1
inkey = scm1
outkey = scm1
include = all
permit = all
qualify = yes
[00:14:22:13:B6:B6]
model = symmetric
host = scm2
inkey = scm2
outkey = scm1
include = all
permit = all
qualify = yes
[00:13:72:4E:D7:54]
model = symmetric
host = sgw2
inkey = sgw2
outkey = scm1
include = all
permit = all
qualify = yes
scm2 will be exactly the same except it has an outkey of scm2. This
should fix your issue with having dundi lookup on the local machine.
I'm not gonna try to understand your ACD stuff right now, so
I'll just
figure you need DUNDi for that and give up on it :) Too busy
fixing the
voicemail app.
--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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[Asterisk-Users] Executing a Function from AGI
Hmmm. Not having much luck with this. I'm trying to call the DUNDILOOKUP
function and assign it to a variable in an AGI script.
I've tried setting with EXEC CMD and with SET VARIABLE. In both cases, it's
treating DUNDILOOKUP literally, rather than calling a funciton.
I've tried this:
EXEC Set DIALPATH=${DUNDILOOKUP(2944093|180net)}
and also:
SET VARIABLE DIALPATH ${DUNDILOOKUP(2944093|180net)}
in both cases, DIALPATH is set to a literal ${DUNDILOOKUP2944093|180net}
What am I doing wrong here?
Doug.
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RE: [Asterisk-Users] Executing a Function from AGI
Python... but it doesn't matter. The examples I pasted where what I am sending to stdout, so the scripting application shouldn't be an issue. -Original Message- From: Alexander Lopez [mailto:[EMAIL PROTECTED] Sent: Thursday, June 15, 2006 3:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Executing a Function from AGI What is you AGI written in?? -Original Message- snip Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Executing a Function from AGI
I'm getting nowhere with this. Is it even possible to set a variable to the
result of a function call in AGI???
-Original Message-
From: Douglas Garstang
Sent: Thursday, June 15, 2006 3:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Executing a Function from AGI
Hmmm. Not having much luck with this. I'm trying to call the
DUNDILOOKUP function and assign it to a variable in an AGI script.
I've tried setting with EXEC CMD and with SET VARIABLE. In
both cases, it's treating DUNDILOOKUP literally, rather than
calling a funciton.
I've tried this:
EXEC Set DIALPATH=${DUNDILOOKUP(2944093|180net)}
and also:
SET VARIABLE DIALPATH ${DUNDILOOKUP(2944093|180net)}
in both cases, DIALPATH is set to a literal
${DUNDILOOKUP2944093|180net}
What am I doing wrong here?
Doug.
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RE: [Asterisk-Users] OPENSER / SER and Asterisk
-Original Message- From: Martin Joseph [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 13, 2006 10:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OPENSER / SER and Asterisk On Jun 13, 2006, at 8:29 PM, Douglas Garstang wrote: If you do this, and not have Asterisk in the call setup path, your going to lose the ability to do a lot of features. What about black/white lists, rate centers, pic codes, intra company extension dialling and other advanced features? Sure, you might be able to do them with SER but good luck trying to find documentation. So, your saying asterisk has better documentation? I just want to be sure I understand you ;~) Absolutely. The SER/OpenSER documentation is terrible, and if you post to the OpenSER mailing list, you get very cryptic replies. ___ --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] OPENSER / SER and Asterisk
Agreed. -Original Message- From: Santosh Rao [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 13, 2006 11:19 PM To: Martin Joseph Cc: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] OPENSER / SER and Asterisk asterisk has a extremely cool documentation. The wiki has everything a newbie like me could hope for.. with samples and everyhting./. where as we are having a very dificult time finding proper documentation or samples and stuff like thtt for SER.. may be if someone good with SER could update ther voip-info/wiki and write some basics abt the ser.cfg or somethjing .. then it would be great. Regards Santosh Rao Martin Joseph wrote: On Jun 13, 2006, at 8:29 PM, Douglas Garstang wrote: If you do this, and not have Asterisk in the call setup path, your going to lose the ability to do a lot of features. What about black/white lists, rate centers, pic codes, intra company extension dialling and other advanced features? Sure, you might be able to do them with SER but good luck trying to find documentation. So, your saying asterisk has better documentation? I just want to be sure I understand you ;~) ___ --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
RE: [Asterisk-Users] OPENSER / SER and Asterisk
-Original Message- From: Jean-Michel Hiver [mailto:[EMAIL PROTECTED] Sent: Wednesday, June 14, 2006 1:47 AM To: Santosh Rao; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OPENSER / SER and Asterisk Santosh Rao a écrit : asterisk has a extremely cool documentation. The wiki has everything a newbie like me could hope for.. with samples and everyhting./. where as we are having a very dificult time finding proper documentation or samples and stuff like thtt for SER.. may be if someone good with SER could update ther voip-info/wiki and write some basics abt the ser.cfg or somethjing .. then it would be great. You can find some very good SER tutorials on onsip.org. You need to subscribe though, but it's free. I haven't read the tutorials, so I could be wrong, but I doubt they'd be very much use. They probably don't do more than give a basic overview, and I'm sure they don't touch things like avpops. ___ --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] DUNDi Docs
Does anyone know where I can find some good DUNDi docs? The ones are dundi.org are absolutely horrible. The examples in dundi.conf are pretty much useless. I still can't figure out why Digium can't write some good documentation. It's their 'baby' after all. This really drives me nuts and pisses people off in general. I've been dicking around with DUNDi for over 6 months and still can't figure it out past the most basic application. Doug. ___ --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] DUNDi Users
I have three Asterisk boxes.
Each has the following in dundi.conf:
180net = dundi_local,0,IAX,dundi:[EMAIL PROTECTED]/${NUMBER},nopartial
180q = dundi_q_pbx1,1,IAX,dundi:[EMAIL PROTECTED]/${NUMBER},nopartial
180q = dundi_q_pbx2,2,IAX,dundi:[EMAIL PROTECTED]/${NUMBER},nopartial
180q = dundi_q_pbx3,3,IAX,dundi:[EMAIL PROTECTED]/${NUMBER},nopartial
My iax.conf on all three Asterisk boxes has this:
[dundi]
type=user
dbsecret=dundi/secret
context=dundi_local
disallow=all
allow=ulaw
allow=g729
I can do a lookup on pbx2 to find where a number is:
hermes*CLI dundi lookup [EMAIL PROTECTED]
1. 1 IAX2/dundi:[EMAIL PROTECTED]/oe_main (EXISTS)
from 00:0e:0c:a1:92:6f, expires in 0 s
2. 1 IAX2/dundi:[EMAIL PROTECTED]/oe_main (EXISTS)
from 00:0e:0c:a1:92:4d, expires in 0 s
DUNDi lookup completed in 53 ms
However, when I dial the DUNDi path, this is what pbx1 logs on the console:
Jun 14 10:51:39 NOTICE[22424]: chan_iax2.c:7215 socket_read: Rejected connect
attempt from xxx.187.142.204, request '[EMAIL PROTECTED]' does not exist
I tried adding the contexts to [dundi] in iax.conf:
[dundi]
type=user
dbsecret=dundi/secret
context=dundi_local
context=dundi_q_pbx1
context=dundi_q_pbx2
context=dundi_q_pbx3
disallow=all
allow=ulaw
allow=g729
However, the call on pbx1 is still routed to the dundi_local context instead of
dundi_q_pbx1.
Do I have to go and modify dundi.conf, so that every dundi entry uses a
different DUNDi user, like this?
180q = dundi_q_pbx1,1,IAX,dundi1:[EMAIL PROTECTED]/${NUMBER},nopartial
180q = dundi_q_pbx2,2,IAX,dundi2:[EMAIL PROTECTED]/${NUMBER},nopartial
180q = dundi_q_pbx3,3,IAX,dundi3:[EMAIL PROTECTED]/${NUMBER},nopartial
And then add users dundi1, dundi2 and dundi3 to iax.conf?
I sure hope not. What a horrible way to have to do it.
Doug.
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RE: [Asterisk-Users] DUNDi Docs
-Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Wednesday, June 14, 2006 9:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DUNDi Docs On Wed, 14 Jun 2006, Douglas Garstang wrote: The examples in dundi.conf are pretty much useless. I still can't figure out why Digium can't write some good documentation. It's their 'baby' after all. This really drives me nuts and pisses people off in general. I've been dicking around with DUNDi for over 6 months and still can't figure it out past the most basic application. What are you trying to do? I am trying to implement distributed ACD queues. A user dials the main queue number 2944000. The primary Asterisk server for that user has 2944000 in it's dialplan. It does a DUNDi lookup of a number, oe_main (it has to be different to 2944000 of course), to determine what the primary asterisk box is for this number, oemain, which is really the ACD Queue. I therefore need to have a DUNDi context that maps to three dialplan contexts. The context are slightly different on each Asterisk server, so that the queue has a primary, secondary, and tertiary server. Like this...: PBX1: [pbx_pri] exten = oe_main,1,Dial(SIP/2944000,20,tr) [pbx_sec] [pbx_ter] PBX2: [pbx_pri] [pbx_sec] exten = oe_main,1,Dial(SIP/2944000,20,tr) [pbx_ter] PBX3: [pbx_pri] [pbx_sec] [pbx_ter] exten = oe_main,1,Dial(SIP/2944000,20,tr) The queue accessed by oe_main is primary on pbx, secondary on pbx2, and tertiary on pbx3. Doug. ___ --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] DUNDi Docs
-Original Message- From: Watkins, Bradley [mailto:[EMAIL PROTECTED] Sent: Wednesday, June 14, 2006 10:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] DUNDi Docs Yes, what is it you attempting? I use DUNDi extensively, though you are correct that the existing docs don't go very far in describing some things. I am trying to implement distributed ACD queues. A user dials the main queue number 2944000. The primary Asterisk server for that user has 2944000 in it's dialplan. It does a DUNDi lookup of a number, oe_main (it has to be different to 2944000 of course), to determine what the primary asterisk box is for this number, oemain, which is really the ACD Queue. I therefore need to have a DUNDi context that maps to three dialplan contexts. The context are slightly different on each Asterisk server, so that the queue has a primary, secondary, and tertiary server. Like this...: PBX1: [pbx_pri] exten = oe_main,1,Dial(SIP/2944000,20,tr) [pbx_sec] [pbx_ter] PBX2: [pbx_pri] [pbx_sec] exten = oe_main,1,Dial(SIP/2944000,20,tr) [pbx_ter] PBX3: [pbx_pri] [pbx_sec] [pbx_ter] exten = oe_main,1,Dial(SIP/2944000,20,tr) The queue accessed by oe_main is primary on pbx, secondary on pbx2, and tertiary on pbx3. Doug. ___ --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] DUNDi Users
It has also just become glaringly apparent to me that a 'reload' does not
always reload the DUNDi configuation.
How can I reload DUNDi without stopping/starting Asterisk?
-Original Message-
From: Douglas Garstang
Sent: Wednesday, June 14, 2006 11:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] DUNDi Users
I have three Asterisk boxes.
Each has the following in dundi.conf:
180net =
dundi_local,0,IAX,dundi:[EMAIL PROTECTED]/${NUMBER},nopartial
180q =
dundi_q_pbx1,1,IAX,dundi:[EMAIL PROTECTED]/${NUMBER},nopartial
180q =
dundi_q_pbx2,2,IAX,dundi:[EMAIL PROTECTED]/${NUMBER},nopartial
180q =
dundi_q_pbx3,3,IAX,dundi:[EMAIL PROTECTED]/${NUMBER},nopartial
My iax.conf on all three Asterisk boxes has this:
[dundi]
type=user
dbsecret=dundi/secret
context=dundi_local
disallow=all
allow=ulaw
allow=g729
I can do a lookup on pbx2 to find where a number is:
hermes*CLI dundi lookup [EMAIL PROTECTED]
1. 1
IAX2/dundi:[EMAIL PROTECTED]/oe_main (EXISTS)
from 00:0e:0c:a1:92:6f, expires in 0 s
2. 1
IAX2/dundi:[EMAIL PROTECTED]/oe_main (EXISTS)
from 00:0e:0c:a1:92:4d, expires in 0 s
DUNDi lookup completed in 53 ms
However, when I dial the DUNDi path, this is what pbx1 logs
on the console:
Jun 14 10:51:39 NOTICE[22424]: chan_iax2.c:7215 socket_read:
Rejected connect attempt from xxx.187.142.204, request
'[EMAIL PROTECTED]' does not exist
I tried adding the contexts to [dundi] in iax.conf:
[dundi]
type=user
dbsecret=dundi/secret
context=dundi_local
context=dundi_q_pbx1
context=dundi_q_pbx2
context=dundi_q_pbx3
disallow=all
allow=ulaw
allow=g729
However, the call on pbx1 is still routed to the dundi_local
context instead of dundi_q_pbx1.
Do I have to go and modify dundi.conf, so that every dundi
entry uses a different DUNDi user, like this?
180q =
dundi_q_pbx1,1,IAX,dundi1:[EMAIL PROTECTED]/${NUMBER},nopartial
180q =
dundi_q_pbx2,2,IAX,dundi2:[EMAIL PROTECTED]/${NUMBER},nopartial
180q =
dundi_q_pbx3,3,IAX,dundi3:[EMAIL PROTECTED]/${NUMBER},nopartial
And then add users dundi1, dundi2 and dundi3 to iax.conf?
I sure hope not. What a horrible way to have to do it.
Doug.
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RE: [Asterisk-Users] DUNDi Users
-Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Wednesday, June 14, 2006 12:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] DUNDi Users If you do a reload pbx_dundi.so, it'll reload the dundi configuration. If you're talking about the strings it returns, if you want to get an immediate result and not use the cache, use something like dundi lookup num bypass. Also, if you have separate entry points for each section of the dundi numbers, you're going to have to have separate users to identify where the call's coming from. If you only use one iax user, you can only use one context. That's like trying to put a phone in two different contexts... where is it supposed to start it's dialing attempts? If you really want, create a context in extensions.conf that includes the other three, because that seems to be the functionality you are attempting. Seems to make sense to me, not sure what's horrible about it :) Ooookay. Why is this possible then? [vmuser] ; Used by voicemail server to authenticate incoming connections username=vmuser type=user auth=rsa inkeys=pbxsys context=vmretrieve context=vmdeposit context=vm_test deny=0.0.0.0/0.0.0.0 permit=xxx.187.142.203 permit=xxx.187.142.204 permit=xxx.187.142.232 permit=xxx.187.142.201 disallow = all allow = gsm I can open up an IAX connection from the client side to any one of those three contexts on the vm system. Why is dundi different? ___ --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] Determining if extension exists
All, Is there a way I can perform a lookup to see if a given extension exists within a given context, on the local system? I could call Dial() and check the result of $DIALRESULT, but I'm thinking there should be a better way. Note, that I don't want to use ChanIsAvail(). That's only for determining if endpoints, ie phones, are registered. It doesn't seem to work with extensions. I am (trying) to use DUNDi. I'd like to perform a local lookup first (seeing as though DUNDi never returns the local system as a path, even when it's valid) before dialling the number via DUNDi if it isn't available. Doug. ___ --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] RE: Determining if extension exists
Worked it out... ChanIsAvail(Local/[EMAIL PROTECTED]) -Original Message- From: Douglas Garstang Sent: Wednesday, June 14, 2006 2:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Determining if extension exists All, Is there a way I can perform a lookup to see if a given extension exists within a given context, on the local system? I could call Dial() and check the result of $DIALRESULT, but I'm thinking there should be a better way. Note, that I don't want to use ChanIsAvail(). That's only for determining if endpoints, ie phones, are registered. It doesn't seem to work with extensions. I am (trying) to use DUNDi. I'd like to perform a local lookup first (seeing as though DUNDi never returns the local system as a path, even when it's valid) before dialling the number via DUNDi if it isn't available. Doug. ___ --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] DUNDi Not Able to Handle Complex Failover Situations
This is driving me nuts. Why doesn't the DUNDILOOKUP function return the weight of a path to a number? The CLI 'dundi lookup' command does. What about the mac address and expiry period? The CLI command returns those, but the DUNDILOOKUP function does not. Why? We absolutely need this in order to perform out routing logic. It has become quite apparent to me that DUNDi is _NOT_ suited to performing failover applications. It is suited to situations where you want to check a number on a series of peers before routing the call through an expensive PSTN gateway. It is not suited to situations where you want to dynamically discover where a number is located within a cluster of Asterisk systems. In our particular scenario, we have ACD queues. Our phones register with a primary Asterisk box. The primary Asterisk box for company A may be different to the primary Asterisk box for company B. In the event that a user in company B wants to reach Company A's queue, we need to use DUNDi to perform a lookup that returns it's company A's primary Asterisk box. However, the primary Asterisk box may have failed, it which case the DUNDi lookup should return the secondary Asterisk system for Company A to the dial plan routing the call. This may have not made sense brain is fried after dealing with this all day. DUNDi seems to be falling really short in performing complex discovery and failover applications like this. If the DUNDILOOKUP fuction returned a weight, it would help a lot. Oh... also when you call the dundi lookup CLI command, you get multiple results. The DUNDILOOKUP function only returns one value. How can I get _all_ DUNDi paths with DUNDILOOKUP? Doug. ___ --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] DUNDi Not Able to Handle Complex FailoverSituations
Who said I was a C programmer? -Original Message- From: Terry Wilson [mailto:[EMAIL PROTECTED] Sent: Wed 6/14/2006 6:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] DUNDi Not Able to Handle Complex FailoverSituations pbx/pbx_dundi.c in dundifunc_read(). shouldn't be too hard to have it set some variables (i.e. DUNDI_RESULT_n) and add the the weight in a CUTable string. Can't return multiple results in a nice manner with the result from a custom dialplan function... I'm working on some other projects right now, but I'm sure the Digium folks would welcome a patch from you if you really want it. On 6/14/06, Douglas Garstang [EMAIL PROTECTED] wrote: This is driving me nuts. Why doesn't the DUNDILOOKUP function return the weight of a path to a number? The CLI 'dundi lookup' command does. What about the mac address and expiry period? The CLI command returns those, but the DUNDILOOKUP function does not. Why? We absolutely need this in order to perform out routing logic. It has become quite apparent to me that DUNDi is _NOT_ suited to performing failover applications. It is suited to situations where you want to check a number on a series of peers before routing the call through an expensive PSTN gateway. It is not suited to situations where you want to dynamically discover where a number is located within a cluster of Asterisk systems. In our particular scenario, we have ACD queues. Our phones register with a primary Asterisk box. The primary Asterisk box for company A may be different to the primary Asterisk box for company B. In the event that a user in company B wants to reach Company A's queue, we need to use DUNDi to perform a lookup that returns it's company A's primary Asterisk box. However, the primary Asterisk box may have failed, it which case the DUNDi lookup should return the secondary Asterisk system for Company A to the dial plan routing the call. This may have not made sense brain is fried after dealing with this all day. DUNDi seems to be falling really short in performing complex discovery and failover applications like this. If the DUNDILOOKUP fuction returned a weight, it would help a lot. Oh... also when you call the dundi lookup CLI command, you get multiple results. The DUNDILOOKUP function only returns one value. How can I get _all_ DUNDi paths with DUNDILOOKUP? Doug. ___ --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
RE: [Asterisk-Users] Asterisk Eyebeam chat function
Unless it's changed recently, Asterik doesn't support the SIP 'MESSAGE' command. Doug. -Original Message- From: Attilla De Groot [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 13, 2006 2:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk Eyebeam chat function Hi all, Eyebeam has a sip-chat function and it would be nice if I would be able to use it. But the problem is that I can't really find information about it. I can just try to send a message and on the Asterisk console a message like this appears: Jun 13 10:05:25 WARNING[6512]: chan_sip.c:7281 receive_message: Received message to sip:[EMAIL PROTECTED] from Bla Sheepsip:[EMAIL PROTECTED];tag=1d072048, dropped it... Content-Type:text/plain Message: ? Can anyone tell me more about this or give me a link with some information about it ? Regards, Attilla de GrootÎ ___ --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
RE: [Asterisk-Users] Asterisk Eyebeam chat function
No problem. SER and OpenSER do support MESSAGE though... -Original Message- From: Attilla De Groot [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 13, 2006 11:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Eyebeam chat function Hi Doug, I didn't knew this. Thank you. Regards, Attilla On Jun 13, 2006, at 4:52 PM, Douglas Garstang wrote: Unless it's changed recently, Asterik doesn't support the SIP 'MESSAGE' command. Doug. -Original Message- From: Attilla De Groot [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 13, 2006 2:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk Eyebeam chat function Hi all, Eyebeam has a sip-chat function and it would be nice if I would be able to use it. But the problem is that I can't really find information about it. I can just try to send a message and on the Asterisk console a message like this appears: Jun 13 10:05:25 WARNING[6512]: chan_sip.c:7281 receive_message: Received message to sip:[EMAIL PROTECTED] from Bla Sheepsip:[EMAIL PROTECTED];tag=1d072048, dropped it... Content-Type:text/plain Message: ? Can anyone tell me more about this or give me a link with some information about it ? Regards, Attilla de GrootÎ ___ --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 ___ --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
RE: [Asterisk-Users] extensions.conf
-Original Message- From: Moises Silva [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 13, 2006 1:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] extensions.conf No limit in code imposed. Not sure about performance penalty for a file that big, have you considered using ARA (Asterisk Realtime Architecture)? On 13 Jun 2006 21:06:52 +0200, andrutto [EMAIL PROTECTED] wrote: Hi Does anyone know how big extensions.conf can be? I am trying to set up Asterisk which will have about 45 000 lines in extensions.conf. Is there any limitation about the amount of lines in that file? Write a perl script that generates a mock 45,000 extensions.conf file, with 45,000 incrementing extensions, throw in a couple of contexts. Start Asterisk and see what happens. Doug. ___ --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] Polycom Queues
Has anyone integrated Asterisk Queues with Polycom phones? What I'd like to do is display the agent status next to their appearance. I don't see much discussion about this. This is not the same thing as setting bw1/bw against the appearance in the phone directory. Thanks Doug. ___ --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] OPENSER / SER and Asterisk
If you do this, and not have Asterisk in the call setup path, your going to lose the ability to do a lot of features. What about black/white lists, rate centers, pic codes, intra company extension dialling and other advanced features? Sure, you might be able to do them with SER but good luck trying to find documentation. -Original Message- From: BILL GITONGA [mailto:[EMAIL PROTECTED] Sent: Tue 6/13/2006 7:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] OPENSER / SER and Asterisk Asterisk does to scale well. Use OpenSER or SER as a front end to asterisk. Make all the sip traffic go through ser and only go to Asterisk for voicemail, IVR i.e media stuff. If you connect to the PSTN using sip, then SER would be used for routing all PSTN calls. --- Erick Perez [EMAIL PROTECTED] wrote: While reading about how to maximize capabilities in asterisk i have read about SER and OpenSER. The sites do not explain to newbies (maybe that's on purpose) what are the benefits of using those products tied with asterisk (or is SER an asterisk replacement??) Can someone give me an idea of what's the usage for open(ser) and asterisk? is it for scalability? should I run it in the same box as asterisk or separated? does it add more functions to asterisk? or is the main function to better handle SIP over firewalls (due to SIP over TCP support)? Thanks for the explanation. -- Erick Perez ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] AGI Stderr
Does anyone know how I can get stderr from AGI to be sent to somewhere other than the console? It seems that this is the only place it can go. Changing logger.conf has no effect. If you want to see errors from AGI scripts, you have to run the Asterisk console, which isn't viable. Doug. ___ --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] AGI Stderr
Oh yeah, I also won't get time/date stamps if I redirect stderr to a file like that -Original Message- From: Douglas Garstang Sent: Monday, June 12, 2006 8:51 AM To: 'Frederic Jean' Subject: RE: [Asterisk-Users] AGI Stderr Frederic, Thanks, but that's not the best approach. I am sending all debug from my AGI script to syslog. I'd like runtime errors to go to Asterisk so that it can log them to a file. If I don't, I'll have files in three places instead of two. (syslog, errors.txt and /var/log/asterisk/*) Doug. -Original Message- From: Frederic Jean [mailto:[EMAIL PROTECTED] Sent: Monday, June 12, 2006 8:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] AGI Stderr Hi Douglas, Try this: open(STDERR, /etc/asterisk/agi-bin/errors.txt) Fred - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, June 12, 2006 11:32 Subject: [Asterisk-Users] AGI Stderr Does anyone know how I can get stderr from AGI to be sent to somewhere other than the console? It seems that this is the only place it can go. Changing logger.conf has no effect. If you want to see errors from AGI scripts, you have to run the Asterisk console, which isn't viable. Doug. ___ --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 ___ --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] Database file to copy for active sessions.
Two solutions... 1. Set OpenSER to to receive registrations from phones. OpenSER 'fans out' the registrations to multiple Asterisk boxes with the send() command. This will break things like call transfer however unless you can guarantee that a transferred call goes back to the same Asterisk box. 2. Write a script that either reads the /var/lib/asterisk/astdb file directly (with DB module), or screen scrapes it with 'asterisk -rx sip show peers'. Pass the registration info to sipsak who sends the registrations to the other Asterisk boxes. Because the registrations come through the normal channels, Asterisk will update what's in memory and what's in astdb with no locking situations. Unfortunately this 'enterprise grade' software doesn't work well in a clustered environment. Doug. -Original Message-From: Jon Schøpzinsky [mailto:[EMAIL PROTECTED]Sent: Friday, June 09, 2006 4:37 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: SV: [Asterisk-Users] Database file to copy for active sessions. There is a solution, but its not straight forward, and not really documented anywhere. A possible solution, is to set a SER server up, before your asterisk, and let that handle the SIP registrations. Jon Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Shenen ShenenSendt: 9. juni 2006 12:21Til: Asterisk Users Mailing List - Non-Commercial DiscussionEmne: Re: [Asterisk-Users] Database file to copy for active sessions. Somy only solution is to use only X-lite softphone where I can add more than 1 proxy, and a Cisco switchboard where I can set up a VRRP protocol, so in case of fall, the cisco make the resolutions of all tables and permited me to call from IP phones like CISCO IP phones or wi_fi phone without problems or registration in asterisk.I think..becouse in this way I see there isn't a solutionright? On 6/9/06, Jon Schøpzinsky [EMAIL PROTECTED] wrote: It's a little more tricky than that. Our solution involves an external manager application, some clever IAX2 routing and dialplan mysql queries. We tried the solution with just copying the registration, but it seems as though the SIP channel has the registry information in an Internal data structure. Jon Fra: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] På vegne af Shenen ShenenSendt: 9. juni 2006 11:56Til: Asterisk Users Mailing List - Non-Commercial DiscussionEmne: Re: [Asterisk-Users] Database file to copy for active sessions. ok...but if I run a softphone and it is registered in the CLI and I see this: -- Registered SIP '655' at 192.168.251.10 port 1175 expires 900 this registration where is put?in which file? Can I copy this registration to another machine? On 6/9/06, Jon Schøpzinsky [EMAIL PROTECTED] wrote: Hello I can save you a lot of time, and tell you that it wont work. It does hold some registration information in the asterisk database, but most of the information is kept internally in Asterisk. Just FYI. Jon Fra: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] På vegne af Shenen ShenenSendt: 9. juni 2006 11:37Til: asterisk-users@lists.digium.comEmne: [Asterisk-Users] Database file to copy for active sessions. How can I copy all the contenent of the asterisk database to another machine? I want copy all the active sessions from one [EMAIL PROTECTED] to another one and running on the second(thisI can do using vrrp protocol, it isn't a problem), I want copy onlyall the active sessions and softphone registrations to another [EMAIL PROTECTED] and then run on it. --No virus found in this incoming message.Checked by AVG Free Edition.Version: 7.1.394 / Virus Database: 268.8.3/359 - Release Date: 08-06-2006 --No virus found in this outgoing message.Checked by AVG Free Edition.Version: 7.1.394 / Virus Database: 268.8.3/359 - Release Date: 08-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --No virus found in this incoming message.Checked by AVG Free Edition.Version: 7.1.394 / Virus Database: 268.8.3/359 - Release Date: 08-06-2006 --No virus found in this outgoing message.Checked by AVG Free Edition.Version: 7.1.394 / Virus Database: 268.8.3/359 - Release Date: 08-06-2006 ___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:
[Asterisk-Users] Polycom subscriptions
Somewhat off topic. We upgraded a Polycom phone from SIP v1.6.3 to v1.6.6 The phone will no longer send SIP subscription messages for buddies to Asterisk. I have broken the directory file down to make it as simple as possible. Here is what it contains. ?xml version=1.0 encoding=UTF-8 standalone=yes? !-- $Revision: 1.2 $ $Date: 2004/12/21 18:28:05 $ -- directory item_list item lnPresley/ln fnElvis/fn ct2944093/ct sd1/sd rt3/rt dc/ ad0/ad ar0/ar bw1/bw bb0/bb /item /item_list /directory I ran a network trace of all traffic to and from the phone on boot. Here is the output of that... 1 0.00 219.187.128.95 - 219.187.142.203 SIP Request: REGISTER sip:ua1.ipt.twoeighty.com 2 0.000101 219.187.142.203 - 219.187.128.95 SIP Status: 100 Trying(1 bindings) 3 0.000148 219.187.142.203 - 219.187.128.95 SIP Status: 401 Unauthorized (1 bindings) 4 0.211291 219.187.128.95 - 219.187.142.203 SIP Request: REGISTER sip:ua1.ipt.twoeighty.com 5 0.211432 219.187.142.203 - 219.187.128.95 SIP Status: 100 Trying(1 bindings) 6 0.237595 219.187.142.203 - 219.187.128.95 SIP Status: 200 OK(1 bindings) 7 3.987556 219.187.142.203 - 219.187.128.95 SIP Request: NOTIFY sip:[EMAIL PROTECTED] (text/plain) 8 4.087355 219.187.128.95 - 219.187.142.203 SIP Status: 200 OK The phone is not sending a SIP subscription message for the watched buddy in the directory. This worked previously in SIP software version 1.6.3. We completely upgraded the sip and phone1 xml files to the ones supplied with sip software 1.6.6 It obviously isn't an Asterisk problem. Doug. ___ --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] RE: IAX Passing Variables
-Original Message- From: Martin Joseph [mailto:[EMAIL PROTECTED] Sent: Thursday, June 08, 2006 1:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RE: IAX Passing Variables On Jun 8, 2006, at 11:04 AM, Douglas Garstang wrote: Well, this kinda sux. What? You repeating yourself ad nauseam? I agree. Yeah, I tend to do that when no one responds. ___ --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] how to identify agi crash cause
I have only seen Asterisk send stdout to the console, which is _extremely_ annoying. If your running a system in production mode, and your having a problem, you have to 1) shut Asterisk down 2) restart the Asterisk console 3) reproduce the problem 4) shut asterisk down again and 5) Restart Asterisk. "enterprise grade".Digium calls it. Doug. -Original Message-From: Josh McAllister [mailto:[EMAIL PROTECTED]Sent: Thursday, June 08, 2006 10:46 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] how to identify agi crash cause STDERR from your agi will be shown on asterisks tty. If youre using safe-asterisk to start, I believe this is redirected to tty9 Or, if you can afford to take asterisk down momentarily, you could just start asterisk without backgrounding it and youll see what your script has to say there. Josh McAllister From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Danish SamadSent: Thursday, June 08, 2006 8:25 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] how to identify agi crash cause Hi,I have a custom agi which at times does not exit gracefull and crashes in between. The logging options are set to the maximum but I dont see something conclusive in the asterisk log.I have noticed it crash after issuing the "SAY NUMBER" and "GET DATA" agi commands and the agi is spawned with no apparent reason after that. I tried running the application locally and debugged but could not reproduce the problem.I also tried enabling core file generation by specifying the following command in /etc/profile "ulimit -c unlimited /dev/null 21" but to no avail, I did not get any core file in /tmp or other locations. Can any one suggest a way to get a core dump of crashing agi's or some other way I can isolate the problem.Any help will be appreciated.Regards,Danish ___ --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] RE: IAX Passing Variables
Well, this kinda sux. We have three Asterisk servers. Phones register to a single, primary server. When a phone on one wants to reach a phone on another, we use DUNDi to discover the destination pbx and IAX to transfer the call to the other Asterisk box. This seems to be a fairly common practice amongst Asterisk users, yes? Well, what about setting variables before call placement? Say you want to set the variable _ALERT_INFO, to have Polycom phones auto answer? Essentially the problem is that channel variables (with the exception of caller id) are not passed from one Asterisk box to another with IAX. How have people gotten around this? Doug. ___ --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] IAX Passing Variables
Well, this kinda sux. We have three Asterisk servers. Phones register to a single, primary server. When a phone on one wants to reach a phone on another, we use DUNDi to discover the destination pbx and IAX to transfer the call to the other Asterisk box. This seems to be a fairly common practice amongst Asterisk users, yes? Well, what about setting variables before call placement? Say you want to set the variable _ALERT_INFO, to have Polycom phones auto answer? Essentially the problem is that channel variables (with the exception of caller id) are not passed from one Asterisk box to another with IAX. How have people gotten around this? Doug. ___ --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] Transcoding g.711 - g.729
I don't know about g.729, but this will work for wav - g711. sox file.wav file.ul Doug. -Original Message- From: Matthew Crocker [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 06, 2006 12:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Transcoding g.711 - g.729 Hello, I have an asterisk server running with 23 g.729 licenses. I have also purchased a sound file from thevoice.digium.com. I need to covert this file (uLaw, PCM I think) to g.711, g.729 g.723 for use with an IVR system. Is there a way I can convert the files using the g.729 digium codec? sox? Thanks -Matt -- Matthew S. Crocker Vice President Crocker Communications, Inc. Internet Division PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com ___ --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
RE: [Asterisk-Users] Polycom SIP 1.6.6
Hi Mark. Thanks...I managed to grab it. -Original Message-From: MBIT Technologies [mailto:[EMAIL PROTECTED]Sent: Tuesday, June 06, 2006 4:32 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Polycom SIP 1.6.6 Send me your details and I can give you a ftp to download it from. Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 024950 5609 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob McKrillSent: Wednesday, 7 June 2006 3:29 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Polycom SIP 1.6.6 I'd suggest calling whoever you buy your phones from. The distributor I work with requires that you are Polycom certified to be able to purchase phones from them, but once you are certified with Polycom you can actually download the firmware from their extranet. On 6/5/06, Douglas Garstang [EMAIL PROTECTED] wrote: Off topic. Anyone know where I can get Polycom SIP software v1.6.6, unofficially?Not that Polycom is analy retentive, or anything like that... Doug___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Asterisk chroot
I thought I saw a guide at voip-info that described how to set up and asterisk to run in a chrooted environment. Now, I can't seem to find it. Anyone know where such a guide may be? Doug ___ --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] RE: Size limitations of extensions.conf
If by database you are referring to an external database, such as MySQL, you have to address failover, redundancy and performance issues if you go in that direction. -Original Message- From: Moises Silva [mailto:[EMAIL PROTECTED] Sent: Monday, June 05, 2006 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RE: Size limitations of extensions.conf Asterisk support the concept of configuration engine, this means that you can write a configuration engine to get the data from anywhere. The default configuration engine is text_file_engine, that reads the configuration from text files. This engine does not have any limit in the code, so the only limit is the performance hit of starting or reloading. Actually some limits exists for the size of context names, nested includes etc, but no for number of lines. Why dont use database engine? instead of large files? Regards On 6/5/06, Brent Torrenga [EMAIL PROTECTED] wrote: If you need to do a couple differing operations on a list of many area/country codes, then you may consider using the database to let the dial plan choose what to do, rather than go through so many extensions. I mention this to keep your extensions.conf easier to read, not because I know whether or not a long extensions.conf will break things... Can someone tell me the size (or any other) limitations for the extensions.conf? We have managed to keep our file pretty small thanks to AGI but we are about to setup a bunch of call restrictions based on area and country code. One line per area code in the US alone adds a LOT of text to this file. Is it a bad thing to have 5 or 6000 lines of text in your extensions.conf on a production system? Will it affect the performance? Sincerely, Brent A. Torrenga Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 tel:+1 219 836 8918 x325 fax:+1 219 836 1138 email:[EMAIL PROTECTED] web:www.torrenga.com ___ --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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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
RE: [Asterisk-Users] Config Revision Control
-Original Message- From: Michiel van Baak [mailto:[EMAIL PROTECTED] Sent: Monday, June 05, 2006 8:03 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Config Revision Control On 09:41, Mon 05 Jun 06, Andrew Kohlsmith wrote: On Saturday 03 June 2006 02:47, Michiel van Baak wrote: I use subversion for this. Every server has its own branch. There's also a branch called 'common' All the server specific branches are svn-copied and svnmerge init from this branche. Then the svn automerge thingie Kevin wrote for the asterisk svn tree is automerging changes to the 'common' tree to all the server trees. In the server trees I make changes specific for one server. Can you give some more details? I am VERY interested in this! Most is already in my previous mail. This is my layout: branches/common branches/servers/home001 branches/servers/home002 branches/servers/cust001 Like that, you get the idea The branches/common holds a full config, cept for sip users etc. So all the [global] and [default] stuff. Also the extensions.conf has some macro's and contexts I need on every machine. The home001 etc hold the conf I actually run on a server. All the specific sip and iax peers/users are defined in it. Also the specific stuff for extensions.conf for that server. If I for example want the congestion in my default outbound routing macro to play congestion for 5 seconds instead of 10 I only alter extensions.conf in branches/common The automerge will take care of the promoting it to all the other branches. Hmmm. What do you do with other files such as AGI scripts, sound files, or music on hold? Do you maintain separate trees for each of these? If you do, to completely update a system, don't you have to check out etc, agi, sound and moh all independantly? Ideally it would be good if you could put it _ALL_ under a single tree, and then put Asterisk in a chrooted envionment. Then you could check out and update the configuration all in one go. While I was playing with svn, it was driving me nuts. It would ALWAYS re-create the current directory, even if I said to check out all files from inside that directory. Means if you went to /etc/asterisk and checked out asterisk, you'd get /etc/asterisk/asterisk. Yuk. Doug. ___ --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] Asterisk chroot
Thanks Patrick, but thats for non-root Asterisk, not chroot Asterisk. Doug -Original Message- From: Patrick [mailto:[EMAIL PROTECTED] Sent: Monday, June 05, 2006 11:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk chroot On Mon, 2006-06-05 at 10:44 -0600, Douglas Garstang wrote: I thought I saw a guide at voip-info that described how to set up and asterisk to run in a chrooted environment. Now, I can't seem to find it. Anyone know where such a guide may be? http://www.voip-info.org/wiki-Asterisk+non-root Regards, Patrick ___ --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
RE: [Asterisk-Users] Config Revision Control
-Original Message- From: Michiel van Baak [mailto:[EMAIL PROTECTED] Sent: Monday, June 05, 2006 8:03 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Config Revision Control On 09:41, Mon 05 Jun 06, Andrew Kohlsmith wrote: On Saturday 03 June 2006 02:47, Michiel van Baak wrote: I use subversion for this. Every server has its own branch. There's also a branch called 'common' All the server specific branches are svn-copied and svnmerge init from this branche. Then the svn automerge thingie Kevin wrote for the asterisk svn tree is automerging changes to the 'common' tree to all the server trees. In the server trees I make changes specific for one server. Can you give some more details? I am VERY interested in this! Most is already in my previous mail. This is my layout: branches/common branches/servers/home001 branches/servers/home002 branches/servers/cust001 Like that, you get the idea The branches/common holds a full config, cept for sip users etc. So all the [global] and [default] stuff. Also the extensions.conf has some macro's and contexts I need on every machine. The home001 etc hold the conf I actually run on a server. All the specific sip and iax peers/users are defined in it. Also the specific stuff for extensions.conf for that server. If I for example want the congestion in my default outbound routing macro to play congestion for 5 seconds instead of 10 I only alter extensions.conf in branches/common The automerge will take care of the promoting it to all the other branches. I use this script to do the automerging every hour: http://svn.digium.com/view/repotools/svn-automerge?rev=54view=markup This also means you have to use the modified svnmerge from the asterisk project: http://svn.digium.com/view/repotools/svnmerge?rev=63view=markup All my servers do auto svn up of the asterisk configs. I guess this is wy beyond my knowledge of subversion. I just started playing with the directory structure I might use, and first thought was something like this: [EMAIL PROTECTED] ~/cfg $ ls -l total 16 drwxr-xr-x 2 dougg users 4096 Jun 5 12:24 acd drwxr-xr-x 2 dougg users 4096 Jun 5 12:28 common drwxr-xr-x 2 dougg users 4096 Jun 5 12:28 pbx drwxr-xr-x 2 dougg users 4096 Jun 5 12:24 vm where acd, pbx and vm refer to a function, or class of systems. pbx/ would have systems pbx1, pbx2 and pbx3 beneath it. Some files, such as sound files, and AGI are common to all systems, and hence the common/ directory. However, I have no idea what to do with it beyond that. I don't know how to push common changes out to all the other servers, or inherit, or whatever, or how to stop a common directory being created on the servers instead of putting the files from common under /var/lib/asterisk/agi-bin and /usr/lib/asterisk/sounds etc. Arrgh. Doug. ___ --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] Polycom SIP 1.6.6
Off topic. Anyone know where I can get Polycom SIP software v1.6.6, unofficially? Not that Polycom is analy retentive, or anything like that... Doug ___ --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] IAX Passing Variables
Well, this kinda sux. We have three Asterisk servers. Phones register to a single, primary server. When a phone on one wants to reach a phone on another, we use DUNDi to discover the destination pbx and IAX to transfer the call to the other Asterisk box. This seems to be a fairly common practice amongst Asterisk users, yes? Well, what about setting variables before call placement? Say you want to set the variable _ALERT_INFO, to have Polycom phones auto answer? Essentially the problem is that channel variables (with the exception of caller id) are not passed from one Asterisk box to another with IAX. How have people gotten around this? Doug. ___ --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] Polycom-Asterisk hints/presence
Oh sweet. -Original Message-From: Rob McKrill [mailto:[EMAIL PROTECTED]Sent: Friday, June 02, 2006 11:25 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Polycom-Asterisk hints/presence According to the release notes for Polycom's SIP 1.6.6 firmware the "Buddy Watch" limitations have been increased from 8 watched buddies to 48 which would give you enough to watch status on three (14 button) side cars. Haven't tested it but read a discussion in the forum about it and plan to test it with a couple of my customers. On 6/2/06, Sean Cook [EMAIL PROTECTED] wrote: Sean, Where did you find that quote, I would like to see the rest of the thread if there was relevant discussions. Thanks.It was really a two email thread... I had sent an email asking what thestatus of BLA/SCA:Here is the entire thread:Sean Cook wrote: I take it SCA/BLA isn't going to make it into 1.4.Anyone have any idea when support will be added to asterisk for this?There has been no BLA support written at this point, and it does notappear that when we do it we will even use SIP-B to get there. SIP-B is very complex (overly so) and doesn't seem like a practical solution forimplementing basic key system type functionality.However... I can say that an implementation of this functionality isbeing worked on at this time, and we intend to make it available in Asterisk as soon as we can. It will most definitely not be in 1.4, but Iwould expect it to appear some time early in the next development cycleand be part of Asterisk 1.6.___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Config Revision Control
Has anyone got any neat solutions for Asterisk .conf file revision control? We have multiple Asterisk boxes here, that we'd like to maintain a _mostly_ common set of conf files on. They aren't all the same though. There's subtle differences. For example,in sip.conf, iax.conf etc, the bindaddr setting is different. Dundi.conf is very different between each system. At the moment I have a file tree on a separate server, and I use the m4 processor to replace certain unique sections of the files. I have a bunch of scripts to build sip.conf etc and then rsync the files out to the servers. It works, mostly, but it isn't elegant. I'd like to revision control all this. I don't know how it could be done with revision control though. As I said, not all the files are the same. I don't know if we'd run a version control client on each Asterisk box, or if we'd run it centrally, and then use rsync again, to copy the files out. Doug. ___ --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] Config Revision Control
But you still have to maintain a completely separate copy for each server by doing that don't you? That's what I am hoping to avoid. It doesn't keep file level versions? Subversion doesn't do that? -Original Message-From: Bruce Reeves [mailto:[EMAIL PROTECTED]Sent: Friday, June 02, 2006 3:03 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Config Revision ControlI setup a subversion server and a trunk for my different server configs. You might look at that, it does not appear to keep file level versions, but it works great here. On 6/2/06, Douglas Garstang [EMAIL PROTECTED] wrote: Has anyone got any neat solutions for Asterisk .conf file revision control? We have multiple Asterisk boxes here, that we'd like to maintain a _mostly_ common set of conf files on. They aren't all the same though. There's subtle differences. For example,in sip.conf, iax.conf etc, the bindaddr setting is different. Dundi.conf is very different between each system. At the moment I have a file tree on a separate server, and I use the m4 processor to replace certain unique sections of the files. I have a bunch of scripts to build sip.conf etc and then rsync the files out to the servers. It works, mostly, but it isn't elegant. I'd like to revision control all this. I don't know how it could be done with revision control though. As I said, not all the files are the same. I don't know if we'd run a version control client on each Asterisk box, or if we'd run it centrally, and then use rsync again, to copy the files out. Doug. ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks ___ --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] Config Revision Control
Title: Message Brad, Not sure if #include statments will help. For that to work, there would have to be a separate directory structure for each server. I'd like to keep it as common as possible. If we had, on our first pbx server... [general]context=frompstn_startallowguest=yes bindport=5060 #include "binaddr.conf" andbindaddr.conf had: binaddr=192.168.10.10 then it's specific to a certain host. It doesn't add any value. I might as well just stick it in the main file. Now, if we could do some sort of variable substition, THAT might work. Doug. -Original Message-From: Watkins, Bradley [mailto:[EMAIL PROTECTED]Sent: Friday, June 02, 2006 3:06 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Config Revision Control The first situation you mention can be solved by creating separate files that contain the unique elements, and then including them in the main files where all the commonality is. That is how we do things, and it works well for us. It may be a little cumbersome if you have a *lot* of uniqueness, but if you really want to share a significant portion of the configs this is the only way I know of to do it. As for revision control, we use Subversion with a branch for each server containing the unique files. All of our configuration scripts also include automatic checkins of changed files (we can always revert if need be). It also makes it easy to spot changes if something goes wrong, as an svn diff will tell you. Regards, - Brad -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas GarstangSent: Friday, June 02, 2006 4:43 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Config Revision Control Has anyone got any neat solutions for Asterisk .conf file revision control? We have multiple Asterisk boxes here, that we'd like to maintain a _mostly_ common set of conf files on. They aren't all the same though. There's subtle differences. For example,in sip.conf, iax.conf etc, the bindaddr setting is different. Dundi.conf is very different between each system. At the moment I have a file tree on a separate server, and I use the m4 processor to replace certain unique sections of the files. I have a bunch of scripts to build sip.conf etc and then rsync the files out to the servers. It works, mostly, but it isn't elegant. I'd like to revision control all this. I don't know how it could be done with revision control though. As I said, not all the files are the same. I don't know if we'd run a version control client on each Asterisk box, or if we'd run it centrally, and then use rsync again, to copy the files out. Doug. =00The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] Config Revision Control
Bruce, Do you run a subversion client on every Asterisk box, and get the files directly, or do run the subversion clienton a single central server, and distrubute them from there? Doug. -Original Message-From: Bruce Reeves [mailto:[EMAIL PROTECTED]Sent: Friday, June 02, 2006 3:03 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Config Revision ControlI setup a subversion server and a trunk for my different server configs. You might look at that, it does not appear to keep file level versions, but it works great here. On 6/2/06, Douglas Garstang [EMAIL PROTECTED] wrote: Has anyone got any neat solutions for Asterisk .conf file revision control? We have multiple Asterisk boxes here, that we'd like to maintain a _mostly_ common set of conf files on. They aren't all the same though. There's subtle differences. For example,in sip.conf, iax.conf etc, the bindaddr setting is different. Dundi.conf is very different between each system. At the moment I have a file tree on a separate server, and I use the m4 processor to replace certain unique sections of the files. I have a bunch of scripts to build sip.conf etc and then rsync the files out to the servers. It works, mostly, but it isn't elegant. I'd like to revision control all this. I don't know how it could be done with revision control though. As I said, not all the files are the same. I don't know if we'd run a version control client on each Asterisk box, or if we'd run it centrally, and then use rsync again, to copy the files out. Doug. ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks ___ --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] Config Revision Control
Title: Message Ok, does anyone know if anyone has already created a guide for using subversion with Asterisk? I've hit a wall already, where the subversion docs say that your files _must_ go into a directory called trunk(huh? What's with that?). That's going to break Asterisk, who obviously wants conf files in /etc/asterisk. Gr. -Original Message-From: Watkins, Bradley [mailto:[EMAIL PROTECTED]Sent: Friday, June 02, 2006 3:06 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Config Revision Control The first situation you mention can be solved by creating separate files that contain the unique elements, and then including them in the main files where all the commonality is. That is how we do things, and it works well for us. It may be a little cumbersome if you have a *lot* of uniqueness, but if you really want to share a significant portion of the configs this is the only way I know of to do it. As for revision control, we use Subversion with a branch for each server containing the unique files. All of our configuration scripts also include automatic checkins of changed files (we can always revert if need be). It also makes it easy to spot changes if something goes wrong, as an svn diff will tell you. Regards, - Brad -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas GarstangSent: Friday, June 02, 2006 4:43 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Config Revision Control Has anyone got any neat solutions for Asterisk .conf file revision control? We have multiple Asterisk boxes here, that we'd like to maintain a _mostly_ common set of conf files on. They aren't all the same though. There's subtle differences. For example,in sip.conf, iax.conf etc, the bindaddr setting is different. Dundi.conf is very different between each system. At the moment I have a file tree on a separate server, and I use the m4 processor to replace certain unique sections of the files. I have a bunch of scripts to build sip.conf etc and then rsync the files out to the servers. It works, mostly, but it isn't elegant. I'd like to revision control all this. I don't know how it could be done with revision control though. As I said, not all the files are the same. I don't know if we'd run a version control client on each Asterisk box, or if we'd run it centrally, and then use rsync again, to copy the files out. Doug. =00The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --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] Config Revision Control
Aaron, I'm trying to check-in (is that the right term?) the files for the first time. There's nothing in the repository yet. Doug. -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Friday, June 02, 2006 3:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Config Revision Control No, if you do an svn co http://svn.server.com/svn/configs/trunk asterisk in /etc, it'll make a folder called asterisk in your /etc directory. Once that's done, any modifications made that are committed to the server can be downloaded into /etc/asterisk by running svn up inside the directory. Might need to get your brakes checked if you keep hitting walls :) On Fri, 2 Jun 2006, Douglas Garstang wrote: Ok, does anyone know if anyone has already created a guide for using subversion with Asterisk? I've hit a wall already, where the subversion docs say that your files _must_ go into a directory called trunk(huh? What's with that?). That's going to break Asterisk, who obviously wants conf files in /etc/asterisk. Gr. -Original Message- From: Watkins, Bradley [mailto:[EMAIL PROTECTED] Sent: Friday, June 02, 2006 3:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Config Revision Control The first situation you mention can be solved by creating separate files that contain the unique elements, and then including them in the main files where all the commonality is. That is how we do things, and it works well for us. It may be a little cumbersome if you have a *lot* of uniqueness, but if you really want to share a significant portion of the configs this is the only way I know of to do it. As for revision control, we use Subversion with a branch for each server containing the unique files. All of our configuration scripts also include automatic checkins of changed files (we can always revert if need be). It also makes it easy to spot changes if something goes wrong, as an svn diff will tell you. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Friday, June 02, 2006 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Config Revision Control Has anyone got any neat solutions for Asterisk .conf file revision control? We have multiple Asterisk boxes here, that we'd like to maintain a _mostly_ common set of conf files on. They aren't all the same though. There's subtle differences. For example, in sip.conf, iax.conf etc, the bindaddr setting is different. Dundi.conf is very different between each system. At the moment I have a file tree on a separate server, and I use the m4 processor to replace certain unique sections of the files. I have a bunch of scripts to build sip.conf etc and then rsync the files out to the servers. It works, mostly, but it isn't elegant. I'd like to revision control all this. I don't know how it could be done with revision control though. As I said, not all the files are the same. I don't know if we'd run a version control client on each Asterisk box, or if we'd run it centrally, and then use rsync again, to copy the files out. Doug. =00The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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
RE: [Asterisk-Users] Config Revision Control
Bruce, But, if you have three servers that function the same, don't you have to check the file out three times and check it back in three times? Doug. -Original Message-From: Bruce Reeves [mailto:[EMAIL PROTECTED]Sent: Friday, June 02, 2006 3:34 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Config Revision ControlI use subversion on a central server and then store each server that is different. The purpose behind it for me was 2 fold, first I have a backup of my configs centeralized and I can roll-back any changes. Second, I can checkout a servers files on a different machine to edit them if I want and check them back when finished. What I meant by file-level is if I edit sip.conf and check it in then the whole svn goes to a new version, not just that file. We use a M$ product that has version control at the file level, so for each file in the library there is a version history. On 6/2/06, Douglas Garstang [EMAIL PROTECTED] wrote: Bruce, Do you run a subversion client on every Asterisk box, and get the files directly, or do run the subversion clienton a single central server, and distrubute them from there? Doug. -Original Message-From: Bruce Reeves [mailto:[EMAIL PROTECTED]]Sent: Friday, June 02, 2006 3:03 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Config Revision Control I setup a subversion server and a trunk for my different server configs. You might look at that, it does not appear to keep file level versions, but it works great here. On 6/2/06, Douglas Garstang [EMAIL PROTECTED] wrote: Has anyone got any neat solutions for Asterisk .conf file revision control? We have multiple Asterisk boxes here, that we'd like to maintain a _mostly_ common set of conf files on. They aren't all the same though. There's subtle differences. For example,in sip.conf, iax.conf etc, the bindaddr setting is different. Dundi.conf is very different between each system. At the moment I have a file tree on a separate server, and I use the m4 processor to replace certain unique sections of the files. I have a bunch of scripts to build sip.conf etc and then rsync the files out to the servers. It works, mostly, but it isn't elegant. I'd like to revision control all this. I don't know how it could be done with revision control though. As I said, not all the files are the same. I don't know if we'd run a version control client on each Asterisk box, or if we'd run it centrally, and then use rsync again, to copy the files out. Doug. ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- BruceNortex Networks ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks ___ --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] Config Revision Control
-Original Message- From: Hadley Rich [mailto:[EMAIL PROTECTED] Sent: Friday, June 02, 2006 3:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Config Revision Control On Saturday 03 June 2006 09:37, Douglas Garstang wrote: Aaron, I'm trying to check-in (is that the right term?) the files for the first time. There's nothing in the repository yet. http://svnbook.red-bean.com That's the documentation that I have been referring to. It isn't particularly helpful. It's example says you MUST have a trunk directory for a start. ___ --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] Config Revision Control
Bruce,
I've been referring
to the book at http://svnbook.red-bean.com/nightly/en/svn-book.html.
The svn book's quick
start says that you must have a trunk directory before you try and import for
the first time.
"For reasons that
will be clear later (see Chapter4, Branching and
Merging), your
project's tree structure should contain three top-level directories named branches, tags, and trunk"
The quick start also does not address how to log in with the
credentials necessary to actually do this...
I get...
svn import /etc/asterisk
svn://216.187.142.202/usr/subversion
Authentication realm: svn://216.187.142.202:3690 example
realmPassword for 'root':
What's the syntax for specifying a user? Is it svn import
/etc/asterisk [EMAIL PROTECTED]://216.187.142.202/usr/subversion???
Doug
-Original Message-From: Bruce Reeves
[mailto:[EMAIL PROTECTED]Sent: Friday, June 02, 2006
3:52 PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [Asterisk-Users] Config Revision
ControlAre you following the quickstart in the SVN book?
For the first time to import them in to a "folder" called trunk. Then as Aaron
stated you can check or co the trunk to any folder.
On 6/2/06, Douglas
Garstang [EMAIL PROTECTED]
wrote:
Aaron,I'm
trying to check-in (is that the right term?) the files for the first time.
There's nothing in the repository yet.Doug.
-Original Message- From: Aaron Daniel [mailto: [EMAIL PROTECTED]] Sent: Friday,
June 02, 2006 3:34 PM To: Asterisk Users Mailing List -
Non-Commercial Discussion Subject: RE: [Asterisk-Users] Config
Revision Control No, if you do an "svn co
http://svn.server.com/svn/configs/trunk
asterisk" in /etc, it'll make a folder called asterisk in your
/etc directory.Once that's done, any
modifications made that are committed to the server can be
downloaded into /etc/asterisk by running "svn up" inside the
directory. Might need to get your brakes checked if you keep
hitting walls :) On Fri, 2 Jun 2006, Douglas Garstang wrote:
Ok, does anyone know if anyone has already created a
guide for using subversion with Asterisk? I've hit a
wall already, where the subversion docs say that your files _must_
go into a directory called trunk(huh? What's with that?). That's
going to break Asterisk, who obviously wants conf files in
/etc/asterisk. Gr. -Original
Message- From: Watkins, Bradley [mailto: [EMAIL PROTECTED]]
Sent: Friday, June 02, 2006 3:06 PM To: Asterisk Users
Mailing List - Non-Commercial Discussion Subject: RE:
[Asterisk-Users] Config Revision Control
The first situation you mention can be solved by creating
separate files that contain the unique elements, and then including
them in the main files where all the commonality
is.That is how we do things, and it works well for
us.It may be a little cumbersome if you have a *lot* of
uniqueness, but if you really want to share a significant portion of
the configs this is the only way I know of to do it.
As for revision control, we use Subversion with a
branch for each server containing the unique files.All
of our configuration scripts also include automatic checkins
of changed files (we can always revert if need be).It
also makes it easy to spot changes if something goes wrong, as
an svn diff will tell you.
Regards, - Brad -Original
Message- From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]
On Behalf Of Douglas Garstang Sent: Friday, June 02,
2006 4:43 PM To: Asterisk Users Mailing List - Non-Commercial
Discussion Subject: [Asterisk-Users] Config Revision
Control Has anyone got any neat
solutions for Asterisk .conf file revision control?
We have multiple Asterisk boxes here, that we'd like
to maintain a _mostly_ common set of conf files on. They aren't
all the same though. There's subtle differences. For
example, in sip.conf, iax.conf etc, the bindaddr setting is
different. Dundi.conf is very different between each system.
At the moment I have a file tree on a separate server, and
I use the m4 processor to replace certain unique sections
of the files. I have a bunch of scripts to build sip.conf
etc and then rsync the files out to the servers. It works,
mostly, but it isn't elegant.I'd like to
revision control all this. I don't know how it could be done with
revision control though. As I said, not all the files are the same.
I don't know if we'd run a version control client on each Asterisk
box, or if we'd run it centrally, and then use rsync again, to copy
the files out. Doug.
=00The contents of
this e-mail are intended for the named addressee onl
RE: [Asterisk-Users] Config Revision Control
Aaron, I followed the quick start guide and created the repository. It'd be really nice if it had some examples of directory structure so I could understand what I am doing. It also doesn't say how to pass the username and password from the svn client. It describes later, sort of, how to create users etc, but doesn't say how to log in, with the quick start guide. Doug. -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Friday, June 02, 2006 3:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Config Revision Control Read this: http://subversion.tigris.org/faq.html#repository http://svn.collab.net/repos/svn/trunk/README That'll link you to the README that comes with subversion, which has a very detailed explanation on how to get a repo set up and running :) If it says anything in there about using trunk, it's just a suggestion. Ours is split out by server name inside a configs folder. On Fri, 2 Jun 2006, Douglas Garstang wrote: Aaron, I'm trying to check-in (is that the right term?) the files for the first time. There's nothing in the repository yet. Doug. -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Friday, June 02, 2006 3:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Config Revision Control No, if you do an svn co http://svn.server.com/svn/configs/trunk asterisk in /etc, it'll make a folder called asterisk in your /etc directory. Once that's done, any modifications made that are committed to the server can be downloaded into /etc/asterisk by running svn up inside the directory. Might need to get your brakes checked if you keep hitting walls :) On Fri, 2 Jun 2006, Douglas Garstang wrote: Ok, does anyone know if anyone has already created a guide for using subversion with Asterisk? I've hit a wall already, where the subversion docs say that your files _must_ go into a directory called trunk(huh? What's with that?). That's going to break Asterisk, who obviously wants conf files in /etc/asterisk. Gr. -Original Message- From: Watkins, Bradley [mailto:[EMAIL PROTECTED] Sent: Friday, June 02, 2006 3:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Config Revision Control The first situation you mention can be solved by creating separate files that contain the unique elements, and then including them in the main files where all the commonality is. That is how we do things, and it works well for us. It may be a little cumbersome if you have a *lot* of uniqueness, but if you really want to share a significant portion of the configs this is the only way I know of to do it. As for revision control, we use Subversion with a branch for each server containing the unique files. All of our configuration scripts also include automatic checkins of changed files (we can always revert if need be). It also makes it easy to spot changes if something goes wrong, as an svn diff will tell you. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Friday, June 02, 2006 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Config Revision Control Has anyone got any neat solutions for Asterisk .conf file revision control? We have multiple Asterisk boxes here, that we'd like to maintain a _mostly_ common set of conf files on. They aren't all the same though. There's subtle differences. For example, in sip.conf, iax.conf etc, the bindaddr setting is different. Dundi.conf is very different between each system. At the moment I have a file tree on a separate server, and I use the m4 processor to replace certain unique sections of the files. I have a bunch of scripts to build sip.conf etc and then rsync the files out to the servers. It works, mostly, but it isn't elegant. I'd like to revision control all this. I don't know how it could be done with revision control though. As I said, not all the files are the same. I don't know if we'd run a version control client on each Asterisk box, or if we'd run it centrally, and then use rsync again, to copy the files out. Doug. =00The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth
