similar to: A problem with monitoring calls

Hello everyone, I can not get the name of the recoding file of agents calls. I set agents.conf as following: ; Enable recording calls addressed to agents. It's turned off by default. recordagentcalls=yes ; ; The format to be used to record the calls (wav, gsm, wav49) ; By default its "wav". ;recordformat=gsm ; ; Insert into CDR userfield a name of the the created recording ; By

Hi all, I have 2 question. I have a call center queue with 5 agent; the following are the configuration files: *queue.conf* [name_of_queue] musicclass = default announce = queue-name_of_queue strategy = ringall servicelevel = 60 context = callcenter timeout = 60 retry = 5 wrapuptime=15 autopause=no maxlen = 0 announce-frequency = 60 periodic-announce-frequency=30 announce-holdtime = yes

Hi, I have the same problem with the queue configuration When I receive 2 calls only 1 phone ring even if more agent's phone are free. The second call will go to an other agent only if the first call is pickup. Somebody have a solution ? This is my config file : Queue.conf [general] ; ; Global settings for call queues ; ; Persistent Members ; Store each dynamic agent in each queue

Hi to all, I have a strange behavior in my asterisk server. I have a queue for 5 agents, the calls enter the queue an go to the agents normally, but if I need to transfer or dial directly to an agent extension that is already in a call, the pbx hung up the actual call (not the transferred call). This is what I see in the log. Called 103 -- Agent/103 is ringing --

Hi, I am moving my call center to 1.4. Previously I was recording calls in agents.conf with the following config recordagentcalls=yes recordformat=wav createlink=yes So I had the filename in all calls which was *connected to agents*. I am looking for a similar functionality for 1.4. I am now recording calls using the following configuration. [general] persistentmembers = no eventwhencalled =

Hello to all I'm trying to record the calls going to my queues, but asterisk creates 2 files, one with the inbound and another with the outbound sound. I know Sox should mix the 2 files automatically in the end, but this isn't happening. I have sox installed in my server. How can I force Sox to mix the files? Here is my config: queues.conf----------------------------- [general]

Hello, I am running asterisk 1.4.22 and Iam recording calls in agents.conf with the following configuration: recordagentcalls=yes recordformat=wav createlink=yes The calls are being recorded , but no entry appears in mysql cdr, and, on the other hand I have other pbx running asterisk 1.2 that do it with the same configuration. In cdr_mysql.conf I have: userfield=1 accountcode=1 Is there a

Hi, I am setting up a test call center using *. I am using one Zap channel (Wildcard TDM400P REV E/F -- 4 FXO modules) for incoming call and sip phones (SjPhone) for call agents. I have setup queues and agents. While testing I found that if the agent presses * key in soft phone while attending calls Zap channel gets hung up, and another customer can call. But if the caller hangs up (for example

after recompilling asterisk (trunk-r75109) after system (mandriva cooker) update (new glibc 2.6, gcc 4.2.1), sound starts very choppy, when codec translation is performed, if translation isn't needed, it sounds OK any idea? until update, everything worked fine. I'm using ztdummy as clock source. during compile, I got lot of errors... ael_main.c: In function ?ast_context_add_ignorepat2?:

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 13579 bytes Desc: image001.png Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20070503/c64c855a/attachment.png

On 2016-08-29 12:28, Eric Klein wrote: > Hi Hooman, > > What you probably want is a PRI PBX running Asterisk. > > You should either plan to build your own (with the cards you need) or get one of the low cost options: > > * Allo.com has their Mega PBX with 1 PPR port (http://allo.com/megapbx-line.html) > * Pika Tech has the Warp PBX with BRI

I am having a problem with createlink not wanting to be disabled in my agents.conf file. No matter what when an agent picks up the phone, it appends the filename. Is there something other than 'createlink=no' that I should be adding to my agents.conf to prevent this? Thanks, Kyle Sexton -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have queue set up in realtime on Asterisk 1.4.2. Below is the senario that is happenening :: I have created a test queue with only one agent. Once I call the test queue the agents phone rings if the aagent is logged on. everything till here is fine. Now if the agent does not pick up the call, the call automaticaly disconnects after 15 secs as set for the queue, till here also it is fine. But

Is anyone successfully using call queues and call groups? If so do you have an example configuration? The wicki and mailing list information I found is pretty old. Thanks! Paul pmahler@signate.com

Hi, Asterisk version : 1.2.7.1 stable version We try agent.conf setting of createlink=yes We always can not see this link value to be filled in MySQL's table filed : userfield But we can see the record file has been created correctly. In debug mode, no userfiled shown in SQL command, May 17 18:10:51 DEBUG[2889] pbx.c: Function result is '"unknown" <2001>' May

Hi To troubleshoot FreeBSD panics triggered by ISDN load on an asterisk system, we are looking to buy an ISDN call generator/simulator device. The minimum requirements include: - Not too expensive - PRI support (BRI support is a plus) - CCS+CRC4 farming + HDB3 coding - EuroISDN (DSS1) support. - A minimum of 4 ports (120 channels/concurrent calls) - Compatibility with Digium cards. - DUT in TE

I am planning to download a large number of files from some website. I am using the following script. files2down = c('aaa', 'bbb', ................) for (i in 1: len) { print(paste('downloading file', i, ' of total ', len)); url = paste(urlPrefix, files2down[i], sep='') destfile = paste (dest, 'inDir', files2down[i], sep='/' )

Hello! FreeBSD tarkhil.titl.ru 6.1-PRERELEASE FreeBSD 6.1-PRERELEASE #5: Tue Mar 14 14:58:53 MSK 2006 root@tarkhil.titl.ru:/usr/obj/usr/src/sys/ARMADA i386 encountered problems with USB-to-IDE box and NEC CD-RW/DVD drive. Controller /dev/usb3: addr 1: high speed, self powered, config 1, EHCI root hub(0x0000), NEC(0x0000), rev 1.00 port 1 powered port 2 addr 2: high speed, self powered,

Hello, How can we monitor agents voice channels for training or quality control purpose. While agent is talking to a customer we need to be able to monitor voice channel (the actual voice conversation). If possible we would like to do that without putting agents in conference rooms. Is there any possible way to do that? Has someone done this? In addition when we tried to put the agent in

"In computer software standards and documentation, the term deprecation = is=20 applied to software features that are superseded and should be avoided.=20 Although deprecated features remain in the current version, their use = may=20 raise warning messages recommending alternate practices, and deprecation = may indicate that the feature will be removed in the future. Features = are=20