similar to: ChanSpy via external application

Hi, I made a simple menu using the Background application and some wav files. I converted the wav files using for a in *.wav; do sox "$a" -r 8000 -c1 "`echo $a|sed -e s/wav//`gsm"; done (from http://www.voip-info.org/wiki/index.php?page=Convert%20WAV%20audio%20files%20for%20use%20in%20Asterisk) The first two files "01/bemvindo" and "01/menu_top" are good.

You got to be kidding about 53 calls being recorded at sametime is an issue. I have done at least twice as many on my dual xeon 3.4Ghz system and had no problem as clients like to record every call that goes through the system. Then again, in my system, the in and out channels are mixed first before they are written to the disk. ________________________________ From:

Hi, I am using Asterisk for a call center on a Dual Xeon machine.. I currently have 109 active channels 53 active calls Every body is complaining about quality and cpu is around 80% idle. Is there any tuning I can do??? Besides that, Asterisk normally goes down once or twice per day... Thank you Dov -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I know this has already been discussed here, but I still have the problem even with 1.2.6: When I call a mailbox in a context "company" is doesn't play my busy message... It goes directly to the temp message... Am I doing something wrong? == Everyone is busy/congested at this time (1:0/1/0) -- Executing NoOp("SIP/200.234.208.250-0840f548", "Voicemail de

Hello, After 2 weeks and 4 days without a problem, Asterisk went down. What happened is that I am using Asterisk 1.2.1 on a machine and have a MySQL for CDR on another machine. The machine with MySQL went down and the Asterisk box was unable to connect to MySQL. This made Asterisk to go down and it was unable to restart until MySQL was back. I know that Asterisk displays a lot of warnings, but

Hi, I got this message on my Asterisk messages file and after it Asterisk went down... 2006-02-18 08:13:55 WARNING[3214] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: + 1 ^ 2006-02-18 08:13:55 WARNING[3214] ast_expr2.fl: If you have questions, please refer to doc/README.variables in the asterisk source.

Hi, For the dial application, parameter g is described as a.. g: When the called party hangs up, exit to execute more commands in the current context. I want the following priority (or at least a priority I can jump to) to be executed anyway, it doesn't matter which party hang up. Is there a way to do so? Thank you Dov -------------- next part -------------- An HTML attachment was

Hi, I am doing an AGI that logs to a database every Agent login/logoff. My idea is to be able to go to this database and check which agents where logged so that I can force their login in case Asterisk goes down for some reason. The problem is that I would need to reload their status from this AGI when Asterisk initializes. Is there a way to do this? One idea I had was to make safe_asterisk to

thank you so much it work you must add 1 like below [app-chanspy] exten => _0071XX,*1,*Macro(chanspy,1234) exten => _0072XX,*1,*Macro(chanspy,5678) exten => _0073XX,*1,*Macro(chanspy,8910) best regards. 2015-03-11 19:48 GMT+00:00 Carlos Chavez <cursor at telecomabmex.com>: > On 3/11/15 12:48 PM, Salaheddine Elharit wrote: > >> hello list, >> >> i use

Hi, Since yesterday my Asterisk 1.2.3 is displaying the following message every few seconds >Asterisk Event Logger restarted >Rotated Logs Per SIGXFSZ (Exceeded file size limit) This causes my log files (verbose, queue_log) to become huge with lots of logger rotate messages, but I don't know which files is exceeding size limit, since even if I delete all log files I still get this

Hi ALL, Is it possible to use symbols # and * in the dialplan for pattern matching? I am doing a "follow me" dial plan, and wanted that my users could dial everything in one shot. But, exten => 888*XXX*XXX,1,NoOp(Follow Me from XXX to XXX) doesn't seem to work... Thank you Dov -------------- next part -------------- An HTML attachment was scrubbed... URL:

hello list, i use chanspy with the code below [app-chanspy] exten => _007.,1,Macro(user-callerid,) exten => _007.,n,Answer exten => _007.,n,Authenticate(1111) exten => _007.,n,ChanSpy(SIP/${EXTEN:3},dqs) exten => _007.,n,Hangup i have a question related to chanspy i have created extension from 100 to 300 and i will give the permission with group of extension i want to use

How can i activate "ChanSpy" to spy on a dedicated extension? I see the following in "/etc/asterisk/extensions_additional.conf" [chanspy] include => chanspy-custom exten => 501**,1,Chanspy(801) exten => 501**,n,Hangup exten => 502**,1,Chanspy(802) exten => 502**,n,Hangup But when i try to call "501**", it doesn't give any response. Thanks.

Hello, I have an E1 and the possibility to use different caller ids in this E1, so, before a Dial, I always have a SetCallerIDNum("User", number). When I check the CDR, the originator of the calls appears to be this "number" I set in the caller id, but not the actual user that originated the call. Is there a way to set a callerid for the outgoing call, but on cdr records to

I'm trying to figure out how to listen in to a channel that I specify. I have the impression I've seen this done via Flash web controls, but I'm trying to write something myself and I can't figure out what command would be used. ChanSpy looks great, but I don't see how to specify the channel. I have a channel identifier like "SIP/provider-08748db0" which is what I

I am still trying to figure out a reasonable way to exit the chanspy application in a dialplan. For the most part I understand how things are working and there is one change I would like to propose. The way the 1.4.23.1 code seems to work is that if there are no channels that match the chanprefix argument the chanspy code stays in a loop waiting for a new channel to come into being that matches

I am trying to find the app ChanSpy for Asterisk v1.0.7. I have tried looking on VOIP-info.org's ChanSpy page (http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+ChanSpy)and also referred to the link regarding bug 3836 (http://bugs.digium.com/bug_view_page.php?bug_id=0003836). I downloaded the attachments and tried to use the patch and compile the source. However, it seems that

Hi list, I have write down my code on which chanspy not working when I make a group with name of spy. Please help me where is the issue on that. a) caller will call this number to join konference and spy group exten => 43681111,1,Answer() exten => 43681111,n,NoOp(****${CHANNEL}****) exten => 43681111,n,Set(GROUP(${CHANNEL})=spy) exten => 43681111,n,Set(a=${GROUP_LIST(spy)}) exten

Hi, I`m trying to use Chanspy for a customer that wants to listen to his employees so he can train them better (or so he claims). In any case, it looks simple but there is something I`m not doing right. When a call is incoming, I set SPYGROUP using Set(SPYGROUP=1234) When I use, on another phone, Chanspy(|qg(1234)) Which should allow me to listen to conversations that hit the first (Set

Hello I have an old Asterisk where I need to listen to Agent calls. So I created this code: exten => _555,1,ChanSpy(Agent) exten => _555,n,Hangup() But I always get: 2009-10-22 16:00:38 WARNING[5695]: pbx.c:1720 pbx_extension_helper: No application 'ChanSpy' for extension (default, 555, 1) It seems that Asterisk doesn't have ChanSpy enabled... is this possible? Which