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...er way. I think you will find the majority of call centers are callback centers. This decision has taken Asterisk out of the realm of providing reasonable call center solutions. VIVA ZAPATA!!! Let's start a revolution! Message: 24 Date: Wed, 20 Dec 2006 22:22:26 +0100 From: Lenz <lenz-ml@loway.it> Subject: Re: [asterisk-users] AgentCallbackLogin() deprecated in 1.4 To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <op.tkvmzo0x3wzjep@pc-lenz> Content-Type: text/plain; format=flowed; delsp=yes; charset...

...el buzzing sounds. My client has such a need because they have analog voice-operated push-to-talk half-duplex devices on the other side, and low level noise from the Asterisk side will keep the channel open. I will try diminishing the TXgain, but I wondered if there were other options too. l. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it

...sh, German, French and Italian. Pages are now encoded as UTF-8. This should make it easy translating QueueMetrics into non-Latin languages. * The "Export as..." icons are not shown in printed versions. A full list of improvements over version 1.3.0 can be found at http://queuemetrics.loway.it/news.jsp You can download the latest version immediately from the downloads page at http://queuemetrics.loway.it/download.jsp , together with the 98-page User manual. As an alternative, if you run RHEL/CentOS/TrixBox/AAH, you can install it automatically using yum - see http://queuemetrics.lowa...

...ave been working lately on a small CDR parsing utility, and would like to do some performance testing on it. I am looking for some - possibly large - real-life Asterisk CDR datasets to run some performance monitoring. Anybody's got some CDRs that can be shared? Thanks in advance, l. -- Loway Research - Home of QueueMetrics http://queuemetrics.com

...Installing the Digium's Asterisk GUI for 1.4 - at http://astrecipes.net/?n=217 It's nothing too complex, but you may find them interesting, especially the new Asterisk GUI. Any comment is welcome - the site is a wiki, so feel free to correct any errors or add improvements. l. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it

Hi, Is it possible to define maximum talk time in a queue? ie any one who joins a queue should not be able to talk more than say 5 minutes to the agent. raj

Hi All, I wonder if any of you has some suggestions on which WebRTC client/softphone to use for a click-to-dial, webpage hosted solution. Any suggestions? Thanks l. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130531/ed3f337a/attachment.htm>

...I was wondering if ther was something more general and that worked whatever your channel configuration might be. I also tried exporting the variable by setting the URL field of the queue() call, but it is not shown in the resulting SIP dialog. Any suggestion but patching the source? l. -- Loway - home of QueueMetrics - http://queuemetrics.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090218/80bd27e6/attachment.htm

...work. What I would like to do would be more on the lines of: Originate call 1 and park it (using a park or waiting) Originate call 2 and bridge it immediately to call1 (using the Application part) But maybe I am missing something? is there anybody who has better suggestions? Thanks l. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130514/33bf16d1/attachment.htm>

...ot seem to be working - is there a way to work around this issue? (I admit this is the fist time I'm trying to use devoce state and the related functions, so maybe there is a very simple slution right in front of my big nose and I'm not seeing it)..... Thanks a lot for your help, l. -- Loway - home of QueueMetrics - http://queuemetrics.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091214/2906cedc/attachment.htm

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...g7ltt at g7ltt.com>) id 1I2zQL-0004tU-PU for asterisk-users at lists.digium.com; Mon, 25 Jun 2007 20:00:54 -0500 As you can see, the message was posted on June 25th and was sent to my email on June 29th! am I the only one who is getting such an awful message turn-around time? l. -- Loway Research - Home of QueueMetrics http://queuemetrics.com

Hi there, Does anyone know how to setup an overflow queue? When a call rings on the queue A, if all agents were busy, the call goes to the queue B. If all agents in queue B were busy, then the call stays on both queues until somebody answers it. I think this is a basic ACD feature available on most PBX that support ACD functionality. Does anybody knows how to do it with asterisk??

Hi there, I support a large number of enterprise users who contractually must connect to our support center via a 4G VOIP connection. I simply want to be able to auto detect all poor quality calls in realtme (as they are being made), play a message and drop the call - without user intervention. All decent call quality calls will be allowed through - to be handled by support staff. Its a

Hi All, I want to track a call that is originated using originate AMI command through AstManProxy server. I m using AstManProxy server and I developed an AstManProxy client. By using my AstManClient program I can able to login AstManProxy server. Now I can able to issue/send originate command to generate a call but I m very confuse that I cannot able to track my call. The AMI events were

Hi, I have these extensions: exten => 101,1,Dial(SIP/101,15) exten => 102,1,Dial(SIP/102,15) exten => 0,1,Dial(SIP/101&SIP/102,15,r) They work fine and I get the ringing sound if I dial them directly. However, I also have this extension: exten => s,1,Answer() exten => s,2,Background(viagenie) exten => s,3,WaitExten() The ringing sound doesn't work for any extension

I have a _very_ specific situation where I need queues to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH

Hi, I'm running 1.2 beta1 in a mini call center. I have 3 queues with 10 operators, and I'm running into some trouble because when all the operators are busy answering call asterisk still sends them more, resulting in a "beep beep" (call waiting) over and over again in Xlite audio. An easy solution woud be the use of a "single line" user agent, like firefly, still

...89621.27|queue-dps|NONE|EXITWITHTIMEOUT|1 I have added asterisks to denote a behaviour I dont understand; the extension 101 is called twice in a row if 103 is unavailable. DO you think this is a bug or there is a valid reason why * behaves like this? (I'm running 1.2.4) Yours l. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it

I'm looking for somehting like the standard house hold linksys/dlink router. Basically it needs to have at least 1x100mbit port, wireless G capabilitys and at least 1 x anolog voip/sip connection. I've found linksys's WRT54GP2 which appears to do what i want. Anybody use this? Does it require vontage's service? I'm looking for any recommendations. Thanks -- ~Shaun