similar to: Delete global variable

Hi there, I've got a small problem with the zaphfc channel. No MSN of an any incoming call which comes trough the ISDN card (Acer ISDN, with HFC chipset and zaphfc driver) which will be forwarded to the SIP-Phone will be displayed. Always it will be shown "asterisk" an the Display. --- snip (zapata.conf) --- [channels] language=de switchtype = euroisdn signalling =

Hello to everybody, does anybody knows how to install Asterisk on a Linksys WRT54G(S)? I had read in the Wiki that it is possible. If somebody has a tip, this would help me very much. Regards Bastian

Hello, I'm searching for a router for our company. Does anybody has a suggestion for a router with a SIP Application Layer Gateway and good working QoS (Upstream AND Downstream). Regards Bastian

My current under-developed project is using rails 1.0 and now plan to upgrade to rails rc1.1,the project uses the Rails-engine and Login-engine as the login module,when I update the rails according to http://weblog.rubyonrails.org/articles/2006/03/22/rails-1-1-release-candidate-1-available The project still can not be started,anybody can tell me the reason?very thanksful!And following is the

Hello *, I try to establish a Asterisk-Server for internal and external usage. Perfect use case for a DMZ, or not? My configuration: I N T E R N E T | | | E | | X | | T |

Hello, I've got problems to install zaptel on a SuSE 9.1 System. The System has got a Linux 2.6.9 Kernel. If I try to load zaptel framework (modprobe zaptel) I get this message: FATAL: Error inserting zaptel (/lib/modules/2.6.9-041214/misc/zaptel.ko): Invalid module format How I can fix this. At compile time, there were no Errors. Regards Bastian

Hello Everybody, I'm using the snom Phones together with Asterisk and I already able to see which Peer is used via "hint" priority. Then a LED on the snom phone is blinking. But I don't see who is calling the other phone. I know that the snom phones are already support this feature. But how I can enable this on Asterisk? Regards Bastian

Hi, i was struggling for a function where only 1 thumbnail should be active and others inactive when clicked on the particular thumbnail. code: 1. view: 2. <script type="javascript"> 3. function PlayVideo_options(videoFile, id) { 4. document.getElementById(id).className = ''video_option active_video''; 5. } 6. </script> 7. 8.

Here''s an alternative in-depth treatment on setting upo rails on Debian (Ubuntu, Kubuntu, ... ), may be interesting to follow through in more detail for some: The Perfect Rails/debian/lighttpd Stack... ( http://brainspl.at/pages/perfect_vps ) Victor Kane http://awebfactory.com.ar On 3/25/06, rails-request@lists.rubyonrails.org < rails-request@lists.rubyonrails.org> wrote: >

Hello Asterisk fans, I try to connect an analog modem to Asterisk. The modems are connected e.g. to alarm systems or a cash terminals (POS). As PSTN-Interface I'm using a Wildcard TE110P (E1). Is it possible to connect the modems to an ATA? Which ATA I should use for that scenarios? Cheers Bastian ____________ Virus checked by G DATA AntiVirusKit Version: AVK 17.1806 from 04.01.2007 Virus

Hi, I tried to use the version 0.6 of chan_capi-cm for outgoing calls it works perfectly but for incoming calls it will not work: --- snip --- *CLI> capi debug CAPI Debugging Enabled -- CONNECT_IND (PLCI=0x101,DID=97,CID=0179903xxxx,CIP=0x1,CONTROLLER=0x1) == reventix: Incoming call '0179903xxxx' -> '97' -- reventix: info element CALLED PARTY NUMBER --

Hi *, I want start with a setup of Asterisk with a clean PC. This PC is a SMP-Machine with two 466MHz CPUs, a Acer ISDN card and a AVM Fritz! PCI card. Which Kernel is better for my constellation (Asterisk with SMP, CAPI and ZAPHFC)? Kernel 2.6.x or Kernel 2.4.x? Regards Bastian

I have an X100P clone hocked up to an analog line of my PRI. I can use it to dial out. but when I call the extension it answers and says "GOODBY" I have a Livevoip DID which successfuly rings to ext 202 I am using asterisk@home and through the AMP inface the line should ring to ext 202 Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf Extensions.conf

Yo all, As there has been some intrest, here's my updated version: I post it to "-dev" as well as "-users", as it may be of intrest to both. Inspired by the example in the tips & tricks-section of "http://www.junghanns.net/asterisk/", I built a more elaborate set of features. Currently, my implementation supports call- forward unconditional, on no answer

Hey, all. I'm trying to set my CID such that, internally, I see a four-digit extension (which is also handy when checking VM), but externally, I see the full 10-digit number. So I plugged these lines into my extensions.conf: exten => _XXXXXXX,1,GotoIf($[ ${CALLERIDNUM} != 1625]?4:2) exten => _XXXXXXX,2,Set(CALLERIDNUM=6031234${CALLERIDNUM:1}) exten =>

Hey, I'm currently using the GotoIf application to set it so if certain caller ID's call my number, it will transfer it to my cell phone, here is the code I have so far. I get an error message that states "call rejected by 198.22.67.70: No such context/extention." when I call the number from my house number. Anyway, here is the code I have. [inbound] exten =>

Has anyone used the H option for Queue() with Callback queues? I want customers in my queues to be able to jump out to voicemail when they get tired of waiting, but in my setup when I pretend to be a customer and press '*' [when I am waiting in the queue] I see the message 'User hit * to disconnect call.' but then just jump out to the outer loop where queued callers wait to

Is what I am trying to do in this context possible. That is changing the incoming CALLERIDNUM. In this case if the incoming CALLERIDNUM is not preceeded by a "1" I want to add a "1". Often calls come in without the preceeding "1" and this plays havoc with my redial if the 3 digit area code matches a local 3 digit extension. All my outside calls are 10 digits or 1+10

I would like to hire someone to automate my asterisk for hosted PBX service for fetures like user signup, adding money and call bridging Please contact me offline at ted at 1ezphone.com ----- Original Message ----- From: "Philipp Kempgen" To: "Asterisk Users" Subject: Re: [asterisk-users] Running AGI script if condition met? Date: Thu, 06 Dec 2007 05:11:24 +0100

Hello Instead of using PrivacyManager, I'd rather use my own dialplan to prompt the user for a ten-digit number if they called while blocking CID. This code does prompt the user, but 1) hangs up if the user didn't type the ten digits before the timeout 2) if the user did type the right number of digits, it still hangs up instead of Returning and then jumping forth to the "cid"