similar to: Dial own extension to get to voicemail.

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

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"

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

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

On Mon, Mar 01, 2004 at 01:25:00PM -0600, oddsock wrote: > the "listener-bandwidth-sharing" aspects of p2p broadcasting has had about > 2 years now to mature. I've talked with people at Abacast, peercast, etc > and they've all said the same thing... "We've got it licked, we have a > viable solution"...however, it's been 2 years now since it all

Hi, I thought I had this problem licked but there still is a rights problem with ARI and Asterisk when using a non-root user (Following the wiki at http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+non-root&diff2=25). When I issue the following: chmod --recursive u=rwX,g=rX,o= /var/spool/asterisk The above command results in the following rights on messages: msg0000.gsm

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

Hello, I have thousands of this messages on my servers ?? Is this a Problem on my site or is the infrastructure broken?? /etc/cron.hourly/0yum-hourly.cron: Could not get metalink https://mirrors.fedoraproject.org/metalink? repo=epel-7&arch=x86_64 error was 14: HTTPS Error 503 - Service Unavailable Thanks for a answer, -- mit freundlichen Gr?ssen / best regards, G?nther J. Niederwimmer

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 >

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, Quick question. Is there a way to have multiple people have an extension, say 900, to their polycom 501 SIP phones on one of the blue buttons to where when a call comes in, I can have it simul-ring and folks can pick up the line on their phone? I'd like to set up a tech support extension for our techs and have a voice mail box assigned to it as well. Right now I just have

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

Adds external weak symbols of constant pool to ExtWeakSymbols set. Lauro -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.llvm.org/pipermail/llvm-dev/attachments/20061220/1aaa0c7b/attachment.html> -------------- next part -------------- A non-text attachment was scrubbed... Name: llvm.patch Type: text/x-patch Size: 1021 bytes Desc: not available

I'm trying to move from SSH1 to OpenSSH 2.9.9p2, under Solaris 8. Initial setup and testing seems to work... including the generation of a new RSA key. The key was created with "ssh-keygen -t rsa" and a passphrase; nothing unusual. I can SSH between machines, both running 2.9.9p2, and debug messages show that this file is being correctly read (I think). It prompts me for the

Hi Oddsock, Like Clement, I am sure Nullsoft is still "offering" AOL's bandwidth since I think Nullsoft is not part of AOL anymore. About the new broadcasting methods, is the multicast technology already available? I have heard only few providers are equipped with multicast enabled routers. What about p2p streaming, is it really reliable? When I see Peercast's statistics, only

I know this has come up before (I checked the archives last night), but just in case someone's got it licked... How can I assure that print jobs from client PCs, through Samba, will print with the username of the account that's logged in, not the guest account or root? We've got 1.9.18p1 running on HPUX 10.20, user security, specific printer definitions created for each printer