similar to: Multiple users and a single extension

Hi all, I've gotten asterisk installed on Debian only to realize that the packaged version is 1.0.7. Is there a reason why they're not up to a 1.2.x release? I'm building a system for production and I'm wondering if I should remain at this old version or if there are any serious issues with 1.2.13 on Debian? Should I be able to do an apt-get from unstable and get 1.2.13 and

Does there seem to be a popular Linux distro folks use specifically for Asterisk? I'd like to move off of FreeBSD but I'm not too familiar with Linux distros. In particular, I'm looking for a free, stable, well supported distro that has a friendly community. Any advice appreciated. Sorry for asking a question that I'm sure has been asked thousands of times. Best regards,

Hey all, I've been doing a lot of playing, and a lot of reading, and it seems people are split as to whereas if they're running their favorite Linux distro and asterisk or Trixbox. I'm getting closer to really looking at a production environment and I'm just looking for any opinions. I'm really enjoying learning linux and asterisk, so initial "ease of use"

I've gotten this Polycom 501 pretty much licked, but I need to know if there's a way in a dialplan to say if someone dials their own extension it goes straight to voicemail and asks them for their password. I thought I saw an example of this on the web but I can't seem to find it. Any advice appreciated! Phil -------------- next part -------------- An HTML attachment was

Hi all, I've got Asterisk 1.2.10 up and running on Debian using the back ports. I noticed that it didn't come with mpg123 or depend on it and I believe I read somewhere that asterisk now handles it's own mp3 playback? Is this true? If so I must have a problem, because I hear no music when putting someone on hold. When looking at the console when putting someone on hold, I see

Hi all, Can someone point me in the right direction here. What I'd like to do with Asterisk is a) dial a 3 digit extention (i.e. 100) on my polycom phones and after the 3rd digit is entered, it dials that extension and b) dial 9 to get out like older PBX systems. Since my internal extensions start with a 1 I think what happens is I enter extension 100 for example, and the phone sits

Greetings, Currently my asterisk box is using Voicepulse. It works fine with the exception that people need to enter the 1+area code for local calls. I'd like to get around this if possible. The following is what I have in my extensions.conf.. exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=6162997590) exten => _1NXXNXXXXXX,n,Dial(IAX2/${VOICEPULSE_GATEWAY_OUT_A}/${EXTEN}) exten

We're moving a somewhat complicated call center over to an Asterisk system, and I'm looking for documentation on queue/agent configuration. So far I haven't found anything on the Digium or Asterisk websites, and I was hoping that someone could point me in the right direction. Thanks, Devon

Hi all, I'm pretty new to linux and compiling modules, but I've scoured the web for help on compiling the zaptel modules from source and I get the following error... make -C SUBDIRS=/usr/src/modules/zaptel modules make: *** SUBDIRS=/usr/src/modules/zaptel: No such file or directory. Stop. make: *** [linux26] Error 2 Any ideas? /usr/src/modules/zaptel is the dir I'm

Hi all, I'm trying to incorporate using the i extension in my callplan to determine if someone enters an invalid extension. My internal extensions are all 3 digits (100-104). The problem is, the callplan doesn't see that say, extension 600 is invalid, it just goes back to the beginning of the callplan and repeats. If I enter a single digit, it works perfectly. Anyone have any

Hi, I am running a 5 seater inbound call center on 1.0.9-BRIstuffed-0.2.0-RC8h and it's running well. I am now trying to upgrade it to 1.2.4. So I installed 1.2.4 from source and copied all config files from original to the new server. But when a caller lands inside the queue no queue message is getting played. The gsm files are present in proper locations, whcih I am able to play using

This possibly isn't 100% asterisk related, but I'd like some opinions/feedback... A customer wanted different ring-tones to differentiate external and internal calls. No biggie once I'd worked out that details - they have 100% GXP2000 phones, so adding in the relevant SIP header and altering the phones to suit seems like it's going to be a solution... But I started to look at

Hi, I want to unsubscribe from asterisk-users-request-lists, and donot want to recieve mail any more. Kindly unsubscribe me... sanchal singh On Mon, 2006-12-18 at 13:57, asterisk-users-request@lists.digium.com wrote: > Send asterisk-users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

Hi all as per the new release notice for 13.33.0 received today - can anyone advise me the max limit of the string to the Dial Command - see * [ASTERISK-27946 <BLOCKED::https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't I have been fighting with this issue for months trying to find a solution I

Hi Dovid Yes was one of the options but as the required list is dynamic becomes very messy - and all combinations problem - where as "call all workers job xxx" is what is needed so the ability to call 20+ numbers is what is needed - agi does a database search for all jobx workers and constructs a dialstring with SIP, DAHDI and Local devices. Can someone tell me where to find maximum

I'm new to Linux, as I've been using Asterisk on FreeBSD via the ports collection. My question is simple - for using the release branch of Asterisk (1.2.13 for now), should I get in the habit of using svn to retrieve the source or should I just download the tarball? Is there a "best practice" or a "recommended" installation method? Thanks in advance, Phil

Any idea what causes the warning "Unable to open pseudo channel for timing... Sound may be choppy."? Any ideas what I need to resolve this? I do have the zaptel module installed but don't have a zaptel card. I'm guessing this has to do with ztdummy? I'm running Debian and installed asterisk, zaptel, and zaptel-source from the backports. Any information appreciated!

Hi all, I'm sure this is a stupid question, but is there a way to check your voicemail by calling your extension from the outside? When I call my own extension from outside and hit pound or star, it just stops my greeting and gives me the "beep". I'd like to call my extension and press a key and be prompted for my password. Otherwise the only way I can think to get around

I am looking for clues on how to configure distinctive ring for a PolyCom SoundPoint 300. Does ALERT_INFO apply? If so, how? Thanks, David Koski david.nospham@kosmosisland.com

If extension 101 calls 102 and user 102 hits # and then 103, the caller ID of 103's phone says 102. I've been looking for a way to have 103's Caller ID show the person that is being transferred not the person transferring. So if my receptionist answers the phone and transfers it to one of my techs, I want my techs phone to display the caller ID of the person who called the