similar to: Queue Limit Callers

Hi, I use asterisk 1.8. My issue is : I have the same SIP members added to two queues. I use realtime configuration and has set the field ringinuse=0 for both the queues. But if an extension is answering the call in one queue, and some new call comes in the second queue, still that extension is ringed. In the queue_log table I am getting RINGNOANSWER events each second for the extension until

Dear friends, ? I am not able to capture the CDR records for unanswered calls. Only one record per call is coming in the CDR table. In the cdr.conf, I have enabled it by setting : unanswered = yes ?

? Hi, From asterisk 1.8, the CDR table is not logging the unanswered or extn busy calls which hit while in the queue. I am talking about this setting in the cdr.conf : ; In brief, this option controls the reporting of unanswered calls which only have an A ; party. Calls which get offered to an outgoing line, but are unanswered, are still ; logged, and that is the intended behaviour. (It also

Hello Folks, I am setting up a call center but we have few agents so one agent is able to handle calls of different languages and different queues. For the agent to identify the caller, I want a popup to appear as the phone starts to ring with the caller's number, language (selected in the IVR), Queue (sales, support etc) and any other information (e.g a URL with parameters) I can send this

Hi friends, Is there any way to prevent an Agent from logging in from a second extension if he is already logged on from an extension. Right now, the scenario is if he login from a second extension, asterisk will automatically log him off from first extension. What I need is that asterisk should tell him that he is already logged on from an extension and should prevent him from logging in

Hi friends, Do we have any way to prevent more than one Agent being logged in from the same extension? Also is there a way to limit an agent from logging in from more than one extension? I searched too much, but didn't find a solution. Please help. Thanks in advance. Shanavaz. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear All, I have six different campaign and 5 different agent have login on that campaign.*Same thing i have done using agi and database,i never use queue management on this scenario. Agent** can also shuffling one campaign to anther campaign. * Now i want to do some work with queue.I want to use single queue to managing this. Eg: campaign Agent Login A a_1,a_3

> Hello All, > > History ~ > I recently took a position with a call center. At the time they had > about 50 agents in a call queue. The queue was setup to ringall. The > agents use Eyebeam softphones. Everything is local lan, no routers, > everything connected via Cisco 3600 10/100 switches. > > Now we are up to about 150 agents, and I have kept everything pretty

Hi friends, I am facing a problem with my asterisk 1.2 PBX. The problem is because of the CLI message "Remote UNIX Connection". After 2 days of a server reboot, this message starts coming. After it starts coming it still works well for few more hours, but then the asterisk hangs. During this time calls are still landing on the system, and calls are going forward upto the queue. Once

Trying to figure out the best way to pull an active call out of a queue by unique id and put it on hold. I don't want to put it on hold on the agent's phone but I want it to be pulled away from the agent's phone and into Asterisk limbo somewhere. Shortly after I want to pull the same call out of limbo and redirect it back to either the same agent or another. I was thinking about call

Hi all, I have been playing with the AMI quite a bit lately - mostly debugging WombatDialer in production, but that's a different story - and I have been frustrated by the lack of a simple way to interact CLI-like with the AMI itself. So I have decided to write something myself to make my life easier, or at least a bit less miserable. The result is a little webapp that you can use as a sort

Hi all, I need some advice - I have been working on originating multiple calls using AMI and then joining them. What I want to do is: - dial call 1 (where the caller is in a "channel" format, like SIp/1234 or Local/1234 at ext) and "park" it somehow - dial call 2 (where again the caller is in channel format) and join it to the previous call. As a requirement, I cannot use the

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:

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

Hello all, I created a set of Docker images running Asterisk and exposing AMI / ARI ports that i found to be quite useful for ARI / AMI development and regression. As they are based on Docker with whaleware, adding new configuration files to roll your own dialplan / queues / voicemail etc is pretty easy. And you can run quite a lot on the same box to simulate clusters. There is no SIP / RTP

Hello List, In PHPAGI, I'm using the Astrisk Manager function send_request() to originate an outbound call. I want to execute the remaining PHP code after the call gets executed (depending on user input). But presently the call originates in a different context and asterisk executes the remaining code in parallel. Is there a way in which I can pause the code execution until the call is

Hello; I am looking for calls recording solution to do recording based on the network traffic .. The solution to be competitive and appreciate if it is open source .. Any suggested one? Regards Bilal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131021/8ed5196a/attachment.html>

We have a queue monitoring application running so we can see the caller ID of callers in a queue. If we see a VIP in the queue, is there any method to force that call to be first in line? If there's a softphone, or queue managing application already written that does this, I'd love to know.

Hi, Will Keypress option will work when am in the queue and hearing MoH? Lets say a caller is waiting in queue and while he is hearing MoH, can he key in some DTMF and go to some other queue? is that possible? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

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