similar to: Install via SVN or tarball?

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,

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

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'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, 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

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 there, this is the first time that I'm building an Asterisk-server. I have compiled Asterisk together with Zaptel on an CentOS 5.3-system. Zaptel is for later, when configuring the POTS-line. Now first internal communication with SIP. Thought it would go easier... I have 2 Grandstream IP-phones : BT-201 and GXP-1200. These are my settings : sip.conf : [root at asterisk asterisk]# cat

I'm trying to compile latest zaptel-1.2.13 and I'm getting following errors: /usr/src/Asterisk-1.2.14/zaptel-1.2.13/xpp/xbus-core.c: In function ?debugfs_open?: /usr/src/Asterisk-1.2.14/zaptel-1.2.13/xpp/xbus-core.c:171: error: ?struct inode? has no member named ?i_private? make[5]: *** [/usr/src/Asterisk-1.2.14/zaptel-1.2.13/xpp/xbus-core.o] Error 1 make[4]: ***

Hi, I fresh installed Fedora Core 6. I downloaded and untar the 'asterisk-1.2.13' into /usr/src/asterisk-1.2.13. When I run ' make' I get: ... ... chan_phone.c:41:29: error: linux/compiler.h: No such file or directory make[1]: *** [chan_phone.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-1.2.13/channels' make: *** [subdirs] Error 1 [root@sss asterisk-1.2.13]#

Dear people, is it possible to have presence using Asterisk 1.2.13 / SIP with Linux/Debian Etch??? I'd like to see if my intranet contacts are available, busy, disconnected.... Thanks a lot Alejandro

Has anyone been able to successfully solve the following issue: WARNING[21725]: channel.c:3024 ast_request: No channel type registered for 'Zap' [Mar 11 01:26:53] WARNING[21725]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) Since we updated asterisk from 1.2.13 to asterisk 1.2.16 the module went away so we updated

Hi Everyone... I am running Asterisk 1.2.13 on Debian "Etch". I installed it from the package. I also installed the web voice mail package, which installed Apache2 and a bunch of other stuff. When I point my browser at my PBX machine, the web page says "It Works!" but of course it does not. It does not seem that Apache is configured to run the vmail.cgi script. In the

hi to all. Do you know if there is an asterisk 1.4 package with h323 support for debian? I've found this http://packages.debian.org/etch/asterisk-h323 but has asterisk 1.2.13. Thanks to all. -- /*************/ nik600 http://www.kumbe.it

Hi All, I've got debian (etch), openvz and asterisk up and running using the openvz wiki guides. The examples use `apt-get install asterisk` and this will install 1.2.13. Has anyone gotten an VPS to compile the latest versions from source? Also, I'm unsure how the zaptel modules come into play, could use some guidance there as well. Thanks. JR -- JR Richardson Engineering for the

I have a customer running Asterisk 1.2.13 with Zaptel 1.2.11 that is having calls dropped. Sometimes you can stay on the phone for a long time and sometimes the call is dropped within a minute. There are 9 lines connected to 3 TDM04B cards. The Panasonic Pbx we replaced did not have this problem. There are 8 SIP phones and 16 analog phones connected to two Astribank-8 units and everyone

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Hi, I'm currently running an * version 1.2.13 and sox version 12.17.5. I want to upgrade sox to the newest release ( 12.18.2 ); need mp3 support. But how do I make the upgrade. Do I need to recompile asterisk afterwards? If I make a " sox -h" after a reboot I can see the new version is running but is that enough? _________________________________________________________________