similar to: FW: [Asterisk-Dev] Strange bells in voicemail and voicemailmain ?

Hi, I've been playing with Voicemail and Voicemail2 a bit for my users, and there are a few things I'm wondering about: - We can specify parameters to the mailbox (s, b or u) to select which prompts to play. However, if we specify 'b' or 'u' it plays that (customisable) message, but it also plays the voicemail instructions. For the dutch, it is customary that a user

Hi, I came back from a quiet weekend today and found my E400P box to have gone astray. Asterisk is loaded from inittab, and started crashing and reloading a couple of thousands of times, each time notifying my monitoring service :-P I remember there would be issues on old cvs stuff since the crash at digium so I made a clean checkout just now. Here is what happens when I load manually:

toy buy my first cisco 186 but when i read this page http://www.djernes.org/~shawn/ata186.htm i cant find in my dev page some parameters just like " UseSIP " what i need to do to show this parameters Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030626/42e66005/attachment.htm

Hi what can be wrong with * that console does not show any stderr text printed from agi script? I am starting with asterisk -vvvvvvvvvvvvvvvvvvvvvvvvrc VERBOSE command does show text on console but printing of STDERR does not I tried it from Perl and from Java and in both cases almost the same result, except in Java more things do not work. In Java for, for example, SAY DIGITS 123 78# would

For anyone that cares... I am porting James Golovich's Manager.pm over to PHP. I plan on also doing some documentation which will cover both the Perl and PHP APIs, which will be almost identical (at least, to whatever extent is practical). Will let y'all know when I have some usable code to show you. -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content &

Hi, Does anyone know of a Print-to-Fax client that works with asterisk & spandsp? Astfax is a partial solution but that only lets us email the fax in, we'ld like to set it up so the user can hit the print button and send the fax (even if all it does is email - transparently to the user - the fax to astfax). -------------- next part -------------- An HTML attachment was scrubbed...

Good Day list I am looking to buy a few Wooksung Video phones to try with my asterisk box.... http://www.wooksung.com/eng/html/pro/pro_001.html has anyone had any experience using these with asterisk? Thanks Ron

Hi, First off, let me state that _YES, I am fully aware that what I am doing is insane, prone to major havoc and bad for general health_ :-)) Scenario: My GF needs an analog modem to use with her banking software (sodding backs don't supply a decent web-application for company use). I am experimenting to see if we can get it to work (albeit slow) trough our ATA186 talking g711 to

Hi all, DIAX 0.9.4 is available for download from the same place: http://www.laser.com/dante or http://www.geocities.com/tdanro The new DLL contain the latest updates made by Steve in the iaxclient library. What's new in 0.9.4: - IAX2 support (new DLL); - selectable DSP: Echo cancellation, AGC, Denoise; - plaintext and md5 authentication supported; - the phonebook is now in a separate

Dear all, I'd like to know if there is a way for multiple asterisk servers to share a common SIP and/or IAX registry. The setup I imagine would be something like : - several asterisk servers called sip1.isp.com, sip2.isp.com, ... - a DNS alias sip.isp.com pointing to all the addresses (thus providing a round robin resolution on each server) - each SIP client would register with sip.isp.com

I'm using DUNDi. My lookup returns 'IAX2' for the tech, and 'dundi:q9sgTFkVMBFdmp0IDX1bYQ@xxx.187.142.204/3254101' for the destination. How do I dial this? I've tried dialling it with: "Dial" "IAX2/dundi:q9sgTFkVMBFdmp0IDX1bYQ@xxx.187.142.204/3254101" passed from my AGI script, but the other endpoint (xxx.187.142.204) is returning: May 11 09:23:41

Hi there, quick question...do digium make any BRI cards (ISDN2) or even better a quad port BRI, maybe im going blind, but I cant see any on their website Cheers Gary

Hi, A potential customer would like to be able to do this: If a call comes in for an employee who is on the phone, allow the front-desk to push the caller in a queue directly to the employee. Now, this is easily done by using queues, but I am curious: What is the performance impact on a system if _every_ employee (phone) has their own queue. How scalable is that in comparison to

Hi, There is any possibility to dial a specific extension and then enter in your own mailbox (the one defined for that specific SIP phone) without asking for the exxtension number but only for the password? I want to be the same extension for all phones, not a specific one for each of them. It is possible to have a time stamp in the recorded message? I want to know when the message has been

Hello all! I have an FXO port on a cisco router that is directly connected to our PBX. Our ATA-186 (firmware version 3) registers with asterisk, which connects to our cisco router's fxo port to give me a dialtone on our PBX from the ATA. How do I pass the flash button to the PBX? It seems the ATA-186 wants to control the flash by putting my first call on hold and prompts me to dial another

Hi, I have an * installation running with chan_capi over a AVM C4 card for quite a while without major hickup. Now I have added a zaphfc device to it and rebuild asterisk with bristuff_0.02RC5. I now experience a lot of drop outs during a conversation. They last 5 seconds and more, but eventually the sound comes back (if the other side has not hang up). Both ISDN cards are currently in the

Hi guys, is there any way to execute a command *after* a caller has hung up the call? Something like exten => s,1,Voicemail exten => s,2,AGI(mwi.agi) I'd like to turn off the MWI on my cellphone (which is done by gammu[1]) Or does anyone know a way to check the state of the MWI from outside, i.e. with a cron-job? I'm turning on the MWI with the email-notify from voicemail, but

iaxclient.sourceforge.net is the home of Steve Kann's crossplatform port of the iax library. iaxComm is a client written in c++ using wxWindows. There is a Win32 binary on the site. I think that it should be compilable on Linux and MacOSX, but can't test it. Feedback is welcome.

I have Asterisk up and running and it is working great with my SIP phones. However, I have some "Skinny"-protocol Cisco 7960s. Does Asterisk support the Skinny protocol? I've seen some references to Skinny in the software. If so, should I stick with Skinny with the 7960 or convert to SIP? If anyone has some Skinny confs they would send me I'd be much obliged. If I should

I've added an important new application: app_transfer. This application is designed to allow Asterisk to request the transfer of an incoming call to a different extension. Consider the following diagram: Caller -> [ PBX1 ] -> SIP or IAX2 -> [PBX2] -> Transfer App A caller calls an extension on PBX1 which forwards to PBX2. PBX2 executes app_transfer, which requests that hte