similar to: telephony that "just works"

Hi, What do I need to do to put an agent into two queues? The idea being that the agent will get the call no matter which queue it comes into? ~ Matt

Hello list, I put together a quick note about how to "see" oh323 calls while they are handled by your * box. http://www.oinko.net/astrecipes/index.php?n=89 The article is just a draft with usage examples; I'd love to hear your comments and updates if there is something I got wrong. Thanks l. -- Assum est, versa et manduca.

When agent tries to transfer a phone call (*2 - att transfer) he hangs up. Why? When a phone call isn't from queue then att transfer works fine. In features conf I have *1 for recording, *2 for att transfer and #1 for blind. In queue blind transfer works. For disconnect I have #0. I guess that * is somewhere defined as for hang-up the call, but where? I can't find it anywhere. Any help

Guys. How do those callcenter apps work with Asterisk where a call comes in and * send a URL and some screen popup up based on callerid or something or username or id and shows all the customers info? Anybody done that? What do you need to do that? If you are using ATAs or IP Phones, how do those integrate with the PC so the screen would popup?

We've been running an Asterisk-based phone system here in our office for a year and a half, and it's pretty much been running smoothly. One employee who works out of the office has a problem that she can't make outgoing calls on a temporary basis every so often (a few times a day). No one else has this problem, her settings are fine, and she regains the capability spontaneously with no

Hello list, I am having trouble listening to GSM files created by Asterisk using a browser. I am noticing that some of my users succeed in listening to them and some others don't. I guess it is something of a codec problem that does not seem to be installed on all machines (though they are all WinXP). Anybody knows what one should do to listen to GSM files? I send files through the

Hello list, I wonder if this is possible with Asterisk: - While talking through Asterisk, I would like a client to start recording a call by typing, say, #99# I know it is possible to do it using an external monitoring application, but I want to know if it's possible to have Asterisk silently monitoring an on-going call and responding to DTMF tones within it. How do things like call

According to http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue, under the "Notes" section: "Transfers of calls that are answered out of a queue must be done using Asterisk '#' transfers (enabled with the 't' option above). SIP transfers result in the Agent remaining affiliated with the call until its eventual termination, preventing that agent from being

Hi, How do I read (make sense of) the timestamp in the queue_log? I'm probably just slow but I don't understand it. Thanks! Regards, Jan

Regarding this article (1) I have one question to make. What can I do to record the call if the agent makes a transfer using the "flash" button instead of "transfer button" or using blindxfer or atxfer defined in features. conf If the agent makes the transfer with "flash", the comunication between the person who is calling and is already in the queue and the target

I know I must be being daft, but is there a way to set which context the queuing system uses when it dials the operators/agents? By default it appears to use the default context. I've looked through voip-info.org and can't find anything, someone please put me out of my misery.

Anyone know how to get app_queue to send logs to MySQL or any other sql server. I found info for cdr's and even configs but nothing on queue_log. If sql is not supported in the current app_queue I will be willing to pay someone to add it. John Bittner Simlab.net

While not asterisk specific related - would be interesting to see something similar done against straight voip calls. You could maybe plot it against e.164.org polls but I think this would be too small a sample set. Can anyone think of something else Asterisk related we could plot this against? Cheers, Dean http://deancollinsblog.blogspot.com/2008/04/skype-24-hour-rolling-analyt ics.html

Does anyone know a softphone that can accept URLs during a call and open that page in the default browser when the call is answered? I Know DIAX and the IDEFISK, only pro version.I need another ones. It can be using the cmd SetURL Regards. -- Bruno de Assump??o Loureiro msn: loureiro_bruno@hotmail.com

I all. I have configured a pair of * servers, sip connected each other Mi problem is the following If on the first * i configure a queue containing phone number of the second * (i.e with a round robin strategy) I have non problem as far as all phones are online. If one of the remote phone number is unavailable, when the round-robin strategy touch that phone the call is answered by the voicemail

Hello list, I have prepared a small tutorial today that deals with how to avoid Asterisk rebuilding DTMF tones when using it to connect industial appliances that use DTMF. You can find it at: http://astrecipes.net/index.php?n=248 I know it isn't everybody's piece of cake, but I thought somebody could be interested as well :) l. -- Home of QueueMetrics -

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

R Experts, I'm looking for some help with the geoR package. I'm trying to krig some data without using a global neighborhood. I would like to set my moving neighborhood to a distance, say 100 meters, where I know my data is spatially correlated. I have tried the ksline function, but that only allows my moving neighborhood to be set to a number of data points. But, since my data

Hello list, as I'm writing a little perl parser for queue_log analysis, I'd like to know *which* telephone answered a specific queue call. Unfortunately app_queue only logs the call id but does not log the call end point. This is okay for SIP endpoints, because their call id is something like SIP/endpointname-1234 so you can reasonably understand who was on answering, but for

Hello list, I have 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