similar to: Asterisk 1.6.0.5 and app_system FAILED using TRYSYSTEM

When I load the asterisk 1.4 gui and log into "/asterisk/static/config/setup/install.html", it tells me "No Analog ports has been detected on your system". I have 2 Wildcard X100P cards that are properly installed. Ztcfg shows no problems. I also get the following message from the asterisk console " app_system.c:105 system_exec_helper: Unable to execute

Hi, I just had my production box deadlocked - no calls could go trough, CLI didn't load. Last lines in log were: [Jan 11 09:15:43] VERBOSE[7265] logger.c: -- Executing [28901 at local_dial:40] GotoIf("SIP/204.11.200.152-c0070ed0", "1?41:57") in new stack [Jan 11 09:15:43] VERBOSE[7265] logger.c: -- Goto (local_dial,28901,41) [Jan 11 09:15:43] VERBOSE[7265]

I have the following problem with the 1.6.0.25 version of Asterisk: 1. A calls B 2. B picks up and talks to A 3. B does attended transfer to C 4. C picks up, but B still hears ringing 5. A and B are connected again (AT timeout exceeded on console) This is exactly the same problem as mentioned in bug 16816 <https://issues.asterisk.org/view.php?id=16816> This bug is solved but filed against

Dear All I want to integrate sugarcrm and asterisk , so when customer call the call center the agent or extension which answers the call , before pickup the phone and talk to customer , view his/her information if it is available. I do this as described below : 1-Setup login username for sugarcrm for each extension 2-Extension Users will login to the sugarcrm. 3-Develop php script to handle new

Hello I didn't find the answer in the ATOF 2nd Ed: When using the Record() application, I need to know how it ended: Did the user leave a message, or did he hang up? If the latter, Asterisk stops right there, while I need to run some other commands before hanging up: ======== exten => _[1-4],n,Playback(/root/asterisk_sound_files/leave_msg) exten =>

Hello About Record(), ATFT 2nd Edition says that "if the filename contains %d, these characters will be replaced with a number incremented by one each time the file is recorded." Problem is, the documentation doesn't explain how to refer to this filename later in the dialplan :-/ In this particular example, I want to move the file to the web server's /htdocs so users can

Hi all, I'm using Asterisk 1.0.9-BRIstuffed-0.2.0-RC8o on box A with a TE410P (EuroISDN cpe) connected to another similar asterisk box B acting as EuroISDN master. I'm performing some load tests by contiously feeding up to concurrent 30 call files to /var/spool/asterisk/outgoing/ on box A which inititate via a dialplan context/extension a outbound call (redirected via chan_local) to

Hi, When I try to run the command wmix to mix two WAV files recorded by the Monitor application I get the following warning in the console and the macro exit at that point. Running the command from a standard system console it works. More, even from this macro it works and produce a valid mixed file, but still get that error and the macro cannot continue. Why? I have tried even with a simple

Well, my faxes passes through asterisk successfully, however I still have some problems about fax reception by rxfax. The softfax answers, and negotiates transmission, however then as some stage of communiation something is wrong. But I have nothing more but this log: Jun 2 00:10:21 DEBUG[16900]: chan_zap.c:4242 zt_read: DTMF digit: * on Zap/10-1 Jun 2 00:10:22 DEBUG[16900]: chan_zap.c:4242

I have a pretty basic Mandrake 10.2 w/KDE 3.2 and I installed Asterisk-1.0.1-2mdk. I installed the source of main and contrib from ftp, so at the install time I accepted all the packages needed to be installed too. The installation went smooth, but when I try to execute asterisk (#asterisk -vvv) I encounter few warnings I end with an error. At this point I didn't touch any conf file, I was

Hi All, This is my first post. I searched the archives and found something similar and I tried some of those suggestions. I changed the file permissions on the scripts directory to 777 (which doesn't seem secure), I also manually ran the detectdahdi.sh script. The response is "None". I am running Mac OS X 10.5.7 with Asterisk 1.4.26.1 which I compiled from source. The Asterisk Gui

I've been working on writing a subroutine to page groups of phones at once and I'm having some difficulty. My goal is to have a user call an extension, I record the page they wish to play, I then page out that recorded file to the phones in groups. [sub-masspage] exten => s,1,NoOP same => n,Answer same => n,Set(filename=$PAGE) same => n,Wait(1) same =>

Hi, I use asterisk branch 1.4 and gui 1.4 as well. I have the following situation: When I try to make gui configuration by http://localhost:8088/asterisk/static/config/setup/install.html I can see that application logs my user correctly but there is no browser window shift to the next page. it stays at the logging one. I get the following info in the console: [Feb 4 09:33:09] == Parsing

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091111/2b828eff/attachment.htm -------------- next part -------------- Hello. I am trying to execute an fax reception script and i am getting the following: [Nov 11 08:40:52] WARNING[12800]: app_system.c:88 system_exec_helper: Unable to execute '/var/lib/asterisk/scripts/mailfax ""

I'm trying to keep the DND status of my Snom phones and the astdb in line but I'm stuck on integrating my gui DND button which talks to * using the manager interface (actually it uses Astmanproxy as the gui host is on a different network to asterisk and can't see the Snom's across the network). All's working fine in my Dialplan; when someone dials the code for DND-on or

Hi, Ive been struggling with asterisk for a few days now. I understand pretty much how it works and how to tie things together (SIP -> SIP internally works fine etc), but just cannot get asterisk to work with an X100P clone (its a Ambient MD3200, if that means anything to you guys). I have tried (initially) asterisk 1.07 with zaptel 1.07, and now Asterisk CVS-HEAD with zaptel cvs. Both give

On this macro I keep getting exited non-zero on s,3, but s,3 is doing what it is suppose to do but the macro stops. Is there a way to make a macro ignore errors and continue to s,4? I have the lattes ver of sox 12.17.4. Also if I just run this line from the command line I don't get an error. [root@redhat monitor]# sox in.wav in-rev.wav reverse [root@redhat monitor]# [macro-record-cleanup]

Hello, Let me start by saying I have checked the wiki and the archives and did find some relative information. I tried the suggestions in those threads, but still have the same problem. I'm using the CVS Asterisk from July 11, 2005. Redhat FC2 SpanDSP 0.0.2pre18 Libtiff 3.5.7 Digium PCI card 1 FXO, 1FXS. I have a single POTS line coming, but I have 2 numbers and am using

Hi All, I must be doing something really stupid as I can't get DAHDi working in Asterisk. It is loaded and working in Linux fine. *CLI> module load chan_dahdi Unable to load module chan_dahdi Command 'module load chan_dahdi' failed. [2010-03-08 14:05:34 CST] WARNING[26926]: loader.c:393 load_dynamic_module: Error loading module 'chan_dahdi': libpri.so.1.4: cannot

Hi all This is a followup on my post "Asterisk 13 - system() dialplan app cannot call bash scripts" from yesterday I've given up trying to use system() to call BASH scripts with parameters from Asterisk 13. Turned out under Asterisk 13.22.0 System() DOES work, but only if you do NOT attempt to pass any parameters to the called script. This works, and reliably calls the script: