similar to: host=dynamic vs host=xxx.xxx.xxx.xxx

Hi all, please forgive me relative lack of knowledge with Asterisk, but I've not played with PBX systems for a while and I'm just re-finding my feet. I've set up my first Asterisk server, I have it configured with a Digium X100P Analogue pots board, I have my Called ID working and everything is hunky dorey with the server itself. But. (and there's always a but) I want to pass

Specifically: exten => s,1,System(/usr/sbin/myperlscript.pl ${CALLERIDNUM}) will execute myperlscript.pl with the caller id as an argument as the first priority. hth -----Original Message----- From: C F [mailto:shmaltz@gmail.com] Sent: Thursday, February 09, 2006 4:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Possible for Asterisk to

Hi So I've setup a Samba Server 3.0 , all working hunky dorey except the creating of users on the fly. Snippet from my smb.conf: passdb backend = tdbsam add user script = /usr/sbin/useradd -m %u delete user script = /usr/sbin/userdel -r %u add group script = /usr/sbin/groupadd %g delete group script = /usr/sbin/groupdel %g add user to group script = /usr/sbin/groupmod -A %u %g delete

Seri, I think you might just need to find the right variable and pass it to your script. Also, you'll probably need to read up on AGI - I'm assuming that's how you would launch your Perl scripts. Here's the list of variables on the wiki: http://www.voip-info.org/wiki-Asterisk+variables Most likely you're looking for the variable referred to as ${CALLERID(num)}

Or you could just use "System" and do it the easy way! AGI is better suited for more complicated scripting. In any case, check out the TFOT book - you'll like it. -MC _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Michael Collins Sent: Thursday, February 09, 2006 3:13 PM To: Asterisk Users Mailing List -

I was stung by this bug yesterday myself and although there are numerous references to this error message I haven't seen any qualified resolution of the problem. Not that the following rant hopes to be such, but it's a good story. My setup is a stable 2.2.8a production domain and an experimental 3.0.2a-SuSE domain. The passwd backend is still smbpasswd. I wish there were a better

Hi there, can anyone shed some light about the use of "host=dynamic" and "defaultip=xxx.xxx.xxx.xxx" in view of iax.conf and sip.conf? In bug 558 I learned that at lest for iax.conf these two settings should NOT be used together. Does the same apply for sip.conf? That would mean that both the Wiki as well as the draft handbook need to be adjusted. Cheers, Philipp

Hi all, I'm experiencing something rather strange with our Asterisk setup. I'll go through the steps. Scenario 1: 1. I call someone on the outside (from my analog phone extension). 2. Everything appears honkey dorey. 3. Transfer called party to conference or another extension - fails. Scenario 2: 1. Receive a call at my extension from the outside. 2. Everything appears honkey dorey.

I am trying to add a copyright disclaimer outside the plot borders of some images I have created. I can use mtext() to add the written portion, but I would like to have the Creative Commons license image (http://en.wikipedia.org/wiki/File:Cc.logo.circle.svg) before the text. I've found that I can plot a .png image inside the plot boundaries using rasterImage() but I can't figure out how to

I looked on google for some time and haven't found a source for 9-911 labels to attach to phones. I have run into them in the past but I can't currently find a source. I am looking for a low volume were having them printed up for me would not be economical. Any suggestions would be welcome. Thanks -Jason -- ----------------------------------------------------------------------------

hi I recently spoke to mr McNamara on IRC, and he mentioned there was a "far better way to do realtime-stuff than the usual realtime in asterisk, and that this was GPL". He failed, however, to ever mention how this could be done, so I just wonder if someone else might know... ? roy -- Roy Sigurd Karlsbakk roy@karlsbakk.net --- In space, loud sounds, like explosions, are even

hey, I bought a license for 729 but I can't use it this is the message. == Registered translator 'g729tolinb' from format 8 to 6, cost 99999 == Registered translator 'lintog729b' from format 6 to 8, cost 18 == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI> WARNING[5126]: File chan_sip.c, Line 1601 (process_sdp): No compatible codecs!

Hi, I installed s fresh system yesterday and it seems that zttool did not install!! ztcfg is there.. Anyone else had this problem or is it just me? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Hi list, I'm interested in receiving incoming call to my Asterisk PBX thru an 1800 number. Anybody knows a provider with best minute rate? I heard that that Nufone can provide this service for around 3cents/min for calls made within 48 continental states. Any provider that can give better rate, even with additional limitation such as much few states that a call can originate? How do

Just wanted to say that everything is working fine, but I may have some suggestions for better documentation once I am done tweaking... Specifically, my server has docker and bridged networking so I ended up with my DNS being populated with some useless IP addresses. Other than that, everything seems to be hunky dorry! Thanks for the great tools!!! Deven

Hi, i have configured asterisk to SIP soft phones. Iam confused about the telephony interfaces like T1,E1,PRI interfaces and how to cnfigure them to asterisk. If iam using a dialogic card in the asterisk server .Can i use this card with a telephone line to dialout to different numebr in the telecom network. SIP-softphone------->---asterisk------>Dialogic card--->telephone line

I receive the following error when I try to call another H323 extension from another H323 when going through *. NOTICE[27669]: File channel.c, Line 1325 (ast_set_read_format): Unable to find a path from 1 to 8 NOTICE[27669]: File channel.c, Line 1296 (ast_set_write_format): Unable to find a path from 8 to 1 WARNING[27669]: File chan_h323.c, Line 528 (oh323_write): Asked to transmit frame type 1,

I am copying from a rsync --daemon (have tried from both W2K/cygwin & Linux daemons) to a W2K machine. Some files get copied every time even though they haven't been touched since the last copy. I have experimented and deturmined that the file size seems to be important. If the file is between 926 & 933 bytes long it will always be copied!! Other sizes and possibly other aspects of

Has anyone had any experience using asterisk for a 911 call center? Does anyone know of any reason why it would not be suitable? As far as I know all 911 call routing takes place at the CO switch so a regular T1 line should work fine. I understand that there is support for ACD in asterisk and that is should be possible to implement screen pop (CTI). Any comments? Gene Kochanowsky Solution

Hi all Asterisk Gurus. I am really badly in need of help. Asterisk is very lovely software, but has one big disadvantage.. lack of documentation.But let's get to the point. 1. Is it normal that I get such a crappy quality with iax, some drops and clicks? Could anyone with some similar setup check my quality and say if this is what the people are so excited about? ( I used to work as a speech