similar to: Determining invalid extensions.

William Warren <hescominsoon at emmanuelcomputerconsulting.com> wrote: >> have you tried trixbox which is an asterisk version based off of Centos? << Seconded. I have an 8-extension, 2-POTS-line + VoIP Asterisk setup here which started life as an Asterisk at Home (precursor to trixbox) install the best part of 2 years ago. It's based on Centos 4.5, and I enabled various

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 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

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

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

Have a question about implementing Call Rollover with my current extensions.conf configuration. [macro-stdexten] exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum exten => s,2,Voicemail2(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce exten => s,3,Goto(default,s,1) ; If they press #, return to start exten =>

Almost two years ago, a change between how AEL code is built into Asterisk dialplan between minor versions made clear the need to provide a sane entry point into AEL subroutines and that's how AELSub() born. With Asterisk 11 release, they way [stdexten] at extensions.conf is invoked changed from Macro to Gosub using the 'missing context feature' and this caused that any stdexten

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Test A: Outside line calling in

Hello all, I'm in the middle of upgrading my asterisk setup to 1.8 (1.8.2.3) and I'm completely confused by the gosub/stdexten thing. I used to call the stdexten macro but I haven't been able to figure out how to use Gosub. I'm using the sample extensions.conf and added something like this: ========================= [home] include => stdexten exten =>

Hello, Asterisk. I am currently using Asterisk (asterisk-1.2.13) and asterisk-addons-1.2.3_1 on FreeBSD 6.1-RELEASE-p10 So, after setup asterisk for realtime extension: res_mysql.conf: [general] dbhost = 127.0.0.1 dbname = asterisk dbuser = asterisk dbpass = asterisk dbport = 3306 dbsock = /tmp/mysql.sock res_odbc.conf: [mysql] enabled => yes dsn => asterisk username => asterisk

Hi, What does the following error mean: Apr 5 12:39:40 NOTICE[22755] app.c: Huh....? no dial for indications? Here is the 'full' log around the error: Apr 5 12:38:24 VERBOSE[22755] logger.c: -- outgoing agentcall, to agent '3002', on 'Local/510@default-6b6c,1' Apr 5 12:38:24 VERBOSE[22755] logger.c: -- Called Agent/3002 Apr 5 12:38:24 VERBOSE[22755]

Hi! Could someone give me a hand? If I dial 200 for echo testing it works... Everytime I dial an extension ex. 505 get the error below.... In this example it was from 508>505 a Xlite Pro to a TA. I believe it has something to do with the way i'm executing the command dial but I use the "standart" that comes in the samples from asterisk. *CLI> -- Executing

I am having a problem with an Asterisk server, in that when it is receiving a call from another Asterisk server using an IAX2 trunk the phone rings for 10 ms and then there is a hungup from asterisk and then the phone rings again before another hangup. The funny thing is that after I really hang up on the calling phone it repeats this as if I am still trying to call. Any Ideas?

Sorry for all the newbie questions. I really appreciate everyone's help today. Okay I've got outgoing and incoming calls working with no echo. yay! Now I'm having an issue with SIP extension to extension calling. Any time I dial another extension it goes right into voice mail. My extensions.conf is pretty small and rough but, here's what I have right now. Most of it was taken

I am running asterisk 1.4 and the Digium GUI SVN-branch-2.0-r4489. When one phone calls another, I see the following on the console (here, 6223 dials 6123) -- Executing [6123 at DLPN_DefaultDialPlan:1] Macro("IAX2/6223-10489", "stdexten|6123|SIP/6123&IAX2/6123") in new stack -- Executing [s at macro-stdexten:1] Set("IAX2/6223-10489",

with the below defined in logger.conf on 11.6 cert 6 I am not getting any log message other than notice and warning in any files when doing module reload logger - queue log is the only one that says it restarts *CLI> module reload logger == Parsing '/etc/asterisk/logger.conf': Found Asterisk Queue Logger restarted built fresh box with make samples - added 2 stations, dialing from

Hi, I'm having problems with transfer from an analog line via a X100p and Cisco 7960's running SIP. With an attended transfer the a call comes in, I transfer it to another 7960, they answer I announce the call, press transfer again, the two parties talk for 1-2 seconds then the analog line drops, though the Cisco phone is not aware of this, i.e. nothing on the screen changes. The

I'm having an issue with CDR. Basically, I expect to have all "legs" of a call having the same linkedid and differing only by the sequence value. That does happen, but I'm getting null dst values after doing an attended transfer. I'm not sure if this is a bug or I'm doing something wrong. I'm running Asterisk 13.2.0. Here's the console log, step by step: First,