similar to: 2 pretty irritating issues....

All, I am trying to set a monitor file from the queue.conf as specified on http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf In order to avoid the default MONITOR_FILENAME format wich is: "agent-xxxxx-uniqueid.wav" for example "agent-10017-1266438575-26.wav" As you may now, when using the queue command you are not able to know which agent will take the call,

The Asterisk Development Team has announced the release of Asterisk 1.8.26.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.26.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 1.8.26.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.26.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

Hi all Does anybody know where this command is supposed to go? Set(MONITOR_EXEC=mv /var/spool/asterisk/monitor/^{MONITOR_FILENAME} /tmp/^{MONITOR_FILENAME}) In the queues.conf file it talks about it. So I naturally thought after I set up my monitor with monitor-format = wav monitor-type = MixMonitor That I could put a lame command in there to convert and move the file elsewhere for backup with

The Asterisk Development Team has announced the release of Asterisk 11.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.8.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 11.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.8.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

Hi I have this in my queue extension and I see this in asterisk when I call to the queue, but no file is created in the directory any ideas? exten => s,1,Set(MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-${UNIQUEID}) -- Executing [s at kundservice:1] Set("SIP/0850001175-b7942770", "MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-1228382046.12") Regards

Hello Asterisk Users, I guess the subject says the most of it; here goes some more detail: - Running Asterisk 1.2.14 - Objective: record all calls managed by a specific queue - Name those files ${TIMESTAMP}-${CALLERIDNUM}-${UNIQUEID} Facts: - If the UNIQUEID chan var is used in the MONITOR_FILENAME, before calling the Queue() application, the two legs of the call are not

Hey Matt I wonder if it is possible that it doesn't work with AEL, does this seem ok to you? s => { Ringing(); wait(2); Answer(); Set(MONITOR_EXEC=/etc/asterisk/lameconvert.php /var/spool/asterisk/monitor/^{MONITOR_FILENAME}); Queue(MyTestQ,ni,,,18); Hangup(); } I have debug

Hi All; I noticed in the queues.conf the configuration for recording the calls in the queuing, and regarding to the filename (or any other parameter), it is written that I can determine the filename using the command: Set(MONITOR_FILENAME=foo) But it should be called from the dialing plan, but really i did not understand how to call it from the dialing plan. Well, for example this is my

The Asterisk Development Team has announced the release of Asterisk 12.1.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 12.1.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New

The Asterisk Development Team has announced the release of Asterisk 12.1.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 12.1.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New

In my * box I've configured two queues and incoming number and whenever any one calls those number call comes to my *box and it sends call to my agents in queue. but if no agent is available it still answer the call. Is there any why when my agents are not available I don't want call to get answered. Here is my dialplan: exten => xxxx,1,GotoIfTime(*|*|20|dec?ccagents,xxxx,6) exten

Hi all, I'm new in the list, and I have a problem upgrading from asterisk 1.2 to asterisk 1.4: There is a diference from asterisk1.2 to asterisk1.4 in AMI events. When I do a call to a queue (with the same extensions.conf dial plan) with ast1.2 and ast1.4, in ast1.2 apper 3 newcallerid event in ast1.4 apper only 2. It is normal? anyone knows it? what is the reason? I

Hi, I've an PRI coming to my asterisk ,calls are coming fine and my agents are able to answer no prob. but I've an agreement with my telco with some incoming no if the no of calls on these no are more then 3 then send to another no. they use busy signal to divert call on another number so I'm sending the call to Congestion() if no of calls in this group are more then 3. But my

Hi everyone I am having problems with my call queue We currently run a customer care call center which has attendants login during the daytime. Customers who call the 'customer care line (a specific number) always get routed to the cutomer care queue (called 124). After hours, staffs of the Network operating center provide customer care services for customers who call in after the last

Hi, Anybody seen this before? (using a pre-compiled asterisk from the OBS on a sles11sp1) (I mean, i did the same with a 1.6 without any problem, but i need 1.8) after starting: kc3004:~ # /usr/sbin/safe_asterisk: line 145: 16133 Segmentation fault (core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY} Asterisk ended with

Guys, if you define recording on queues.conf and also define a monitor_filename var on your dialplna, you can record a queue call but, isthere a way to do something with the file after the call ends? I need to move the file to some other place but I cant find where to define a command to run after a queue call finishes. Any hints?

Hi, I'm sure some else has been facing this problem. I want to record all the call coming in my queue. I want this format: YYYYMMDD-HHMMSS-AgentID-CallerId - UniqueID. I'm using the monitor feature inside the queue.conf. I can't use the agents.conf monitor features because I'm using dynamic agent (addqueuemember) The problem I'm facing is that I can

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, I'm trying to use CEL to display channel information in real time. It works fine for simple calls, blind transfers, or SIP attended transfers (initiated from the SIP phone). My problem is for Asterisk attended transfers (atxfer as configured in features.conf). The scenario is: . phone 107 calls phone 100, . 100 dials the atxfer code,