similar to: Disconnect supervision in India?

What happens if you take out the Zaptel I/F's? If it boots, you can correct whatever you did then replace them. hth -----Original Message----- From: Chris Earle (CBL) [mailto:cearle@cbltech.ca] Sent: Tuesday, February 28, 2006 7:45 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cannot boot machine up after working on zaptel.... Hi all, hard for me to explain this, but

Hi all, Has anyone ever thought about or implemented a way to use Rails to pull content from a database and actually *generate* static html pages from view templates? I want to throw a bunch of content into a db using rails and then put all my website templates into the views dir and write the files out with the content inside. The funny thing is that the html pages Rails normally outputs to

Does anyone test Asterisk 1.2.X + bristuff-0.3.X and TDM card? We can't get it to work. -David -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Julian J. M. Sent: Friday, March 31, 2006 1:44 AM To: Chris Earle; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: BRI

Hi all, getting a server going wiht a few TDM400's and some phones, and some IAXys too I haven't heard any issues about AU phones being able to RING in Australia, like the problem in the UK with ring capacitors on the BT system. Are there any problems like that? Also, with the iaxy's -- they should work (and ring) in Australia right? The only hint I'm seeing around is the use

This extension http://meta.wikimedia.org/wiki/Auto_Login_via_REMOTE_USER requires web server authentication right? Correct me if I'm wrong, but this means that the contents of the User Table would have to be in the passwd file defined through the .htaccess file right? .. because it passes whatever the user authenticates on the webserver with through to the wiki/extensions scripts right?

Hi all, is it possible to bridge a call between a Junghanns quadBRI card and a TDM400 in the same server? It should be I think, -- I am trying this and when an incoming call comes in, it hangs both up at the moment the bridge is attempted (and a subsequent 'qozap: dropped audio' error is show in the /var/log/messages) Any thoughts appreciated -- I've seen posts, but no clear

Is it possible to run the CVS-HEAD/Stable version of Zaptel (1.2 whatever) with an older version of Asterisk? I'm running 1.09, but I was wondering if I could get at the newer echo cancellers like KB1 and MG2 without upgrading to Asterisk 1.2? I'm going out on a limb here to try and fix a serious echo problem on a TDM + BT PSTN line in the UK Thanks for your suggestions everyone --

Hi all, general query here --- I'm about to set up an asterisk box for use in Japan but can't figureout if it's all ISDN there or what? I have gathered so far that the two major providers, NTT and KVH both offer ISDN lines with .......INS1500 and maybe INS64 protocols? Not sure... But I'm seeing stuff about J1 vs. T1/E1 .... so does that mean I can't use a Digium card it

> > > Hi, > > > I checked by changing to from-zaptel, but no luck yet. Pls guide me on > this. > > Regards, > vudura senadeera > > > ------------------------------ > > > > Message: 9 > > Date: Fri, 19 Jan 2007 16:47:18 -0000 > > From: "Robert Jenkins" < raj@jrw.co.uk> > > Subject: RE: [asterisk-users]

Hi all, I'm trying to get a junghanns QuadBRI to coexist in the same machine as a Digium TDM400P card (so I can run the ISDN lines in and bridge with analog phones plugged into the TDM). I'm having a problem loading the modules. If I follow the BRIstuff (0.3.0-pre-1l) install method.... it's to modprobe zaptel, then insmod qozap.o I'm on Debian 2.4.31. That works. But then I

I'm a complete Asterisk novice and have an installation based on the Asterisk@HomeCD. I've installed my TDM400P with 2 x FXO & 2 x FXS, but every time I try to dial out, I get a message "No circuits available". Can someone confirm the pinouts for connecting the FXO's to a UK BT Line - I have RJ11 connectors on the back of my TDM400P card, so ideally I'd like to

Thanks. I think our problem ca be similar. Have you tried to call from analog phone #1 to another analog phone #2? It works. But when you try to call vice versa from #2 to #1 it does not work. When you restart asterisk it works again - but only one direction. -David ________________________________ From: asterisk-users-bounces@lists.digium.com

Hi all I'm using sangoma a200 cards in the UK and have the ongoing, often noted problem of disconnect supervision with BT POTS lines. Just noticed this post on http://www.voip-info.org/wiki/view/UK+Asterisk+Details stating that potentially someone's got a solution : "TDM400P &amp; Not Detecting Hangups: Got a TDM400P installed and having problems with Asterisk not detecting

Arrrrgh I hate trying to sift through all these messages and keep track of the various threads going on ......... Who else on here prefers the newsgroup/threaded approach? If you haven't already, check out news.gmane.org for mailing lists turned into newsgroups readable by news readers....... only problem being that this list requires list membership before

Hey all, quick question: does asterisk work okay in a Cygwin environment? I want to install it on my cygwin setup for local testing/demoing and save me the hassle of using a pure linux machine........ As long as it doesn't take a huge huge performance hit from running out of Cygwin, then I'll have a go there for a start confirmation appreciated! thanks -- C h r i s E a r

Hi, Is it possible to monitor conversation of logged in Agents? Currently I am using ZapScan to monitor incoming calls, but I would like to monitor individual agents. raj

Hi , Thanks danny nicholas. Finally we get the things done with following. If i specify busypatten=500,500 then asterisk does not recognize hang up signal. After removing it only all are working fine. I choosed 2nd option as per your suggestions. working chan-dahdi.conf: ==================== signalling = fxs_ks busycount = 3 busydetect = yes callprogress = yes progzone=in usecallerid=yes

Hi, Is it possible to define maximum talk time in a queue? ie any one who joins a queue should not be able to talk more than say 5 minutes to the agent. raj

Hi, I have a typical call center with queues and agents added via AddQueueMember. One of my requirement is to implement a forgot password function. If a caller does not remember the password, he calls up an unauthenticated line and the agent manually authenticates him. Then the caller should have a provision to reset his password. The requirement is that the agent should not know the new password

Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digium card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over IAX trunk from which calls get routed to third server (C) (1.6.0.9) again via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is where