similar to: Returning values from macro inside Dial command

Hi: I had an fxo card from Digitnetworks and it was working fine on my Asterisk box. I then replaced it with TDM04B. I changed the zaptel and zapata to include the four channels. When I run ztcfg, I get configuration errors: Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04:

I am having a problem getting my files backed up to the location I want to from a WinXP client going to a Debian rsync server. The following is sending asm/ to /home/robert/backup folder. rsync -avz /cygdrive/d/asm robert@192.168.1.102:backup/ My understanding of the rsync man page is that this is supposed to send asm/ to what the path that is defined in rsyncd.conf, mine is defined as /backup.

i dont how to edit the time for ringing "30000ms" to "40000ms" when it displayed on console "Nobody picked up in 30000 ms" and its very short time for ringing . please if anyone can help me do it please. ____________________________________________________ Sell on Yahoo! Auctions ? no fees. Bid on great items. http://auctions.yahoo.com/

hello i am trying to develop perl application for asterisk with radius accounting how can i debug that weather callback is working when call is stoped. how can i check this syslog('info', 'hello Asterisk!'); thanks Kamran ____________________________________________________ Sell on Yahoo! Auctions ? no fees. Bid on great items. http://auctions.yahoo.com/

I have just setup a Linux box (Suse 9.0) to act as an internet gateway and file server. The installation has gone very well, but I am having a problem with my Samba server. (I have a small network with about 4 client computers, all Windows boxes.) Problem 1: I set "Share Homes" in the Samba configuration, but none of the user folders or even the Home directory could be seen from the

hello how to solve these errors /var/lib/asterisk/agi-bin/agi-rad-auth.pl line 10 use Asterisk::AGI; vi /etc/asterisk/extensions.conf exten => _X.,1,agi,agi-rad-auth.pl|Routing=SIP&AuthorizeBy=SIP vi /etc/asterisk/modules.conf load => res_agi.so <---------------errors------------------------> *CLI> Can't locate Asterisk/AGI.pm in @INC (@INC contains:

Hi all! We are using 1 Samba PDC and 2 bdc (Version 3.0.15pre3-SVN-build-UNKNOWN-PS-SuSE) with openldap2-2.2.6-37.38 on SLES 9. New users setup ok and first logon password change works. Because of HIPAA we need the passwords to change every 30 days however this isn't happening. I thought that I had this working once upon a time while I was testing and getting ready for production but

A call comes in and goes into the queue, the queue dials a sip channel using a macro. The macro plays a set of options to the callee and if the callee presses 3 it sets MACRO_RESULT=CONTINUE and the macro ends. For some reason the caller goes back into the queue rather than continueing on in the dial plan. Why is this, i could have sworn in 1.2 if i set MACRO_RESULT=CONTINUE that the

This is a followup to the posting earlier about Hunt Groups with Call Screening. I have implemented the following macro and for some reason the Macro does not exit and continue the context it was called from when the called party rejects the call - It always drops through to the NoOp at the end and connects the call. Below are two examples of the dial commands I am using to call the macro.

Hi, Does anybody knows how to get this work? I have been searching all over and can?t find the way. Using chan_sccp tge phone can log in and has line tone. But the problem is that it is not able to receive nor dial out. Any help would be apprecieted. Thanks Javier --------------------------------- Sell on Yahoo! Auctions - No fees. Bid on great items. -------------- next part

I am trying to use a macro to screen calls by calling several different phones at the same time in different groups. Find me will not work and queues will not work either. Trust me, I've tried them both and they don't work like they should. Here is what I have: A call comes into 6084 and does the following (in default context): exten => 6084,1,NoOp(test) exten =>

Does anyone know if the CVS repository is on-line? I am getting: NEW CVSROOT: :pserver:anoncvs@xiph.org:/usr/local/cvsroot (password authentication) cvs login (Logging in to anoncvs@xiph.org) cvs [login aborted]: connect to xiph.org:2401 failed: Connection timed out *****CVS exited normally with code 1***** Thanks for any help. JOnathan. __________________________________________________ Do

Cher Areski, Je m'excuse si je me suis tromp? sur vos intentions. Cependant j'aimerais souligner que je ne suis pas le seul a avoir trouv? l'installation de votre application un peu trop compliqu?e et les instructions du "Idiots Guide" impr?cises. Il faut dire que Linux n'est dej? pas facile pour un ancien de Microsoft comme moi. Par cons?quent lorsque les instructions

I am setting up an after-hours on-call system. Someone calls in and requests service, and while they listen to music on hold, we dial out to several people's cell phones and home phones. We don't know if they will be answered by the employee, or by voicemail or a spouse/relative/child/pet. So we play a message that says "press 1 to accept the call" and ask employees to train

I am following the example at http://www.voip-info.org/wiki/view/Asterisk+tips+findme but I find that no matter what, the call is connected. Can anyone confirm that config is working for them? Any suggestions appreciated. I need to transfer calls to a list of cell phones, ring all of them, allow them to screen the call, connect the call to the first number that accepts the call, and allow

Hello list, I'm using the following macro when calling an external callphone/GSM number : [macro-press1] exten => s,1,NoOp() exten => s,n,Playback(/var/lib/asterisk/sounds/prompts/press1) exten => s,n,Read(INPUT,,1,1,1) exten => s,n,NoOp(input : ${INPUT}) exten => s,n,GoToIf($["${INPUT}"=="1"]?exit:hangup) exten => s,n(exit),NoOp(call accepted) exten

Hello, What i am trying to do is ..... Dail a number and ask if you wana talk to XXX press 1 and if you dont wana talk press any other key. For this purpose i am using this link<http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial> . *I am using this option :- * *M(**x**)*: Executes the macro (x) upon connect of the call (i.e. when the called party answers). IMPORTANT - The CDR

Hi list, We're trying to set up a phone number that customers can call to get a hold of anyone of a group of sysadmins (and not their voice mails!). We found the findme example ([1]) that makes the callees press 1 to accept the call. It almost works, but it doesn't work correctly when one of the callees, the sysadmins, hangs up after accepting the call. We're using this

I'm trying to screen the call transfert to my cell phone using a exemple found on the web. http://www.voip-info.org/wiki-Asterisk+cmd+Dial It work partially: while I'm prompted to accept the the caller still her the music. But wether I accept the call or reject it I'm put in communication with the caller. I can see in the log that MARCO_RESULT is set to CONTINUE when I reject the call

Hi all, I am using Amanda and smbtar for backing up Windows machines. However, in addition to the errors I get on open files (which is to be expected), I am getting errors on files that should be fine to access. Here is a snippet from the Amanda log which contains smbclient errors: ? code 0 opening remote file \ebsco-1064\cueanalysisnew.zip (\ebsco-1064\) ? Expected SMBtrans2 response, got