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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:

..._________________ 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 and cpu is...

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, 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 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 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

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

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, 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

...k-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 > >> > >...

Hi! A few months ago I needed some help for the following issue: .) a call comes in .) Person A takes the call and does an attended transfer to Person B .) Person A hangs up the phone without waiting for Person B taking the call .) the caller get lost at this point !! At this point the attended transfer should go into a blind transfer. The phone of Person B should still be ringing and the

Hi ALL, I have downloaded app_chanspy.c and chanspy_sounds.tgz. But I haven't found any instructions on how to compile and where to untar these files... I tried to put the .c file on <asterisk-src>/apps and remake asterisk, but it seems it was not enough... 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 was only able to ChanSpy Agent channels. How do I monitor outgoing calls? Thank you Dov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060119/a7eee094/attachment.htm

Hi, I want to count the number of open Zap channels on my server. [outgoingzap] exten => _0NXXXXXX,1,NoOp(Outgoing Local - 7 digs - ${EXTEN:1}) exten => _0NXXXXXX,2,Set(ZAP01=$[${ZAP01} + 1]|g) exten => _0NXXXXXX,3,Set(UPDATED=true) exten => _0NXXXXXX,4,Dial(${TRUNK}/${EXTEN},60) exten => _0NXXXXXX,6,Busy exten => _0NXXXXXX,7,Playback(thank-you) include => hangupcontext

Hi, I had a wct1xxp in my asterisk server, but I migrated to a cisco sip gateway, and then unplugged the e1. I then changed zaptel's Makefile to include ztdummy and ran modprobe ztdummy Music on hold for queues is not working well... it is simply mute. I realized that, while waiting on a Queue, if I ran a reload, the music on hold starts being played for a few seconds and then stops, until

Hello, Today I had again problems with CDR. My MySQL cdr table was corrupted and thus CDR couldn't be logged. At this moment Asterisk console started to display the following message "Avoided deadlock for '0x843fa98', 10 retries!" hundreds, thousands of times (together with the table corrupted message), until it simply displayed a "Terminated" message and went

Hi, On the general section of my sip.conf I had a disallow=all. Then I put disallow=all, allow=g729, allow=ulaw on my users. It didn't work until I removed the disallow=all from the header. I know disallow=all in the header is totally useless in this case (since I have it for every user), but anyway, is this the correct behavior? Thank you Dov -------------- next part -------------- An

Hi, In features.conf I have defined "atxfer => 1" So, when a customer calls my support queue, and the agent from my support queue needs to transfer the customer to the billing queue, the agent dials 1, hears a "transfer" message and then dials the billing queue extensions. The agent enters a queue. At this point, he can hang up and leave the customer in the queue. But

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