similar to: A problem with recoding agents calls via monitor

Hello everyone, I have a problem with getting name of the recorded file of agent calls. As I've googled I found that the name of the recording file should be inserted in userfield of CDR table. To do this I set createlink=yes in agents.conf but still userfield of cdr is empty but the recrding file is created. my agents.conf file is like this(That part related to recording options): ; Enable

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 =

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 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 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'm working with asterisk on a project and I found a problem with cdr_odbc. As we know, after answering each call a cdr event is raised which is saved in cdr_csv and cdr_odbc. but here my point is on cdr_odbc. some information, including start_time and end_time is given by cdr event but the problem is that these two information(start_time and end_time) is not getting save in cdr_odbc. I

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

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?:

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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

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:

Hi I have noticed that asterisk returns 'SIP 603' when the called party does not answer. My test setup is simple: two SIP phones (extensions: 100 and 111) registered to an Asterisk 1.8.30.0 gateway.The Dial timeout is 30 seconds. When 100 calls 111 and after 30 seconds, asterisk sends a CANCEL request to 111 (expected) and a '603 Decline' response to 100 (unexpected &

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]

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

Hello, I am using Queue application with multiple agents in each queue. I want to set the CDR(userfield) for each cdr based on the agent answering the call. Is it possible to do this? Thanks

Hello, I'm using Asterisk with Realtime extensions and ODBC CDR. And I'm have some trouble with the CDR userfield that is not changed when using the SET command in the realtime dialplan. In my dialplan (extensions.conf, the file) I'm setting the userfield like this : exten => s,n,Set(CDR(userfield)="X") Later, my dialplan switches to the realtime part and this is an

Hi, I'm trying to migrate from Asterisk 1.8 to Asterisk 13 and can't figure this one out. I'm pretty sure the question has been already asked, but I failed to find a solution. Can you modify CDR values in an h-extension? My cdr.conf contains: [general] enable=yes unanswered=yes endbeforehexten=yes initiatedseconds=no batch=no The diaplan contains a simple "h" extension

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