search for: bigio

...nfo.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. But the third file (01/menu_top), fails in the end of the sentence, and this message "Auto fallthrough, channel 'SIP/dov.bigio-ae4a' status is 'UNKNOWN'" appears in the console. -- Executing Goto("SIP/dov.bigio-ae4a", "01.menu.locaweb|s|1") in new stack -- Goto (01.menu.locaweb,s,1) -- Executing Answer("SIP/dov.bigio-ae4a", "") in new stack -- Executi...

...o' (language 'pt') == Spawn extension (macro-ramais_sip, s, 224) exited non-zero on 'SIP/200.234.208.250-0840f548' in macro Here are the "show voicemail users for company" results Context Mbox User Zone NewMsg company 87 Dov Bigio 0 Thank you Dov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060404/0d6b7995/attachment.htm

...and be able to monitor the call. My idea was to, when the user clicks on the Agent, I would Originate a call between his extension and the extension I have for ChanSpy, passing as parameter the Agent number. For testing this, I tried a call file on /var/spool/asterisk/outgoing Channel: SIP/dov.bigio -----------> This is me MaxRetries: 3 RetryTime: 40 WaitTime: 25 Context: 01.telecom Application: ChanSpy Data: Agent/5450 ---------> This is the Agent I want to monitor Priority: 1 The problem is that ChanSpy doesn't accept "Agent/xxxx" as par...

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 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, 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.

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, 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, 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, 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

...___________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Matt Roth Sent: Tue 4/11/2006 5:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call center running Asterisk - sound quality- critical! >>On 4/10/06, Dov Bigio <dovb@terra.com.br> wrote: >> >>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 an...

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 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:

Hi all, I have agents who are members of more than one queue. When an agent is busy with queue A, he is not considered busy by queue B, and receives call (since his EyeBeam Softphone has 6 channels). Besides that, I use a monitoring tool that connects through the manager interfaces and run "show queues" and "show agents" to know agents statuses. I need Asterisk to consider

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 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 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

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

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