similar to: queus & agents

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 have developped an application that monitors the status of my queues through the events triggered on the Manager Interface. This way, I can know the status of my Agent real time. Now, I have a new requirement that I must allow a manager to click on the Agent he wants to monitor and be able to monitor the call. My idea was to, when the user clicks on the Agent, I would Originate a call

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.

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, 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 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

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.

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

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, 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

Hi, I have the following requirement.. after a customer is answered by a Queue, I want him to be redirected to another extensions, where an IVR would answer and ask for his opinion about the analyst who just solved his issue. Is there a way to redirect him automatically, or do I have to ask my agents to manually transfer the users to this IVR extension? Thank you Dov -------------- next part

Hi, I have several queues configured at my call center for different support levels. Today, something weird happened: - A client called queue 1 and was answered by an agent - The agent transferred the call to an user (not a queue), by dialing the atxtransfer (1) key defined in features.conf - The user transferred the client to another Queue, by using the second channel and the XFer key of her

Hi, I wanted to use the same extensions for Pausing and UnPausing queue members. Is that a variable that is set up with the agent status (on call, available, not logged, paused) so that I could use it to make some logic in this extension? exten => 111,1,Set(AGENTEPARADESLOGAR=${$[AGENTBYCALLERID_${CALLERIDNUM}]}) exten => 111,2,PauseQueueMember(|Agent/${AGENTEPARADESLOGAR}) exten =>

Hi, I am making a Dialplan in which I have to record some outgoing calls from some users and not from the others. I made to outgoing calls Macros, when that records calls (using Monitor()), and another that doesn't: [macro-outgoinglocal] and [macro-outgoinglocalrecord] Is it possible to have a user defined variable in sip.conf, so that before calling a Macro I can check the value of this

Hello, I have installed SlimServer for Windows on my desktop and Asterisk on a Red Hat Linux machine. I am able to play mp3's for music on hold when mp3s are on the Linux server, and to play streaming mp3's with Windows Media Player and Winamp on Windows using the slim server. I also have mpg123 on my Linux, apparently installed correctly, since it works for local moh. I put the

Hello, is it possible to change how are caller id on incoming call from isdn, capi lines displayed od sip phones ? ( e.g. SNOM ) standard is 1234567@domain.net. I just want only 1234567 to be displayed. is it possible ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/

Hi, I am using 1.2.3, and sometimes I can see the following message: Mar 2 08:42:42 WARNING[25937] 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 ^ Mar 2 08:42:42 WARNING[25937] ast_expr2.fl: If you have questions, please refer to doc/README.variables in the asterisk source. Any ideas? Thank you

Hi, I am doing some tests with Asterisk's ACD capability, and as far as I could go I have realized that each agent defined in agents.conf must keep a session (call) open with Asterisk in order to be considered online. When a user calls, the agent receives a beep notification in his softphone and he answers to the pending call in the open channel and after the call ends he remains on the open