similar to: Queue time to answer/abandon + OrderlyStats Server Edition.

Hello Voipcrazy, It's funny you should mention that - we've just released (as in today) a FREE version of our OrderlyStats service for call centre and queue monitoring and management. OrderlyStats features realtime (synchronous/message-based) display of all queue, agent and caller events so you can see what's happening in your call centre as it happens. The control panel feature

Hi, Is anyone here using OrderlyStats with asterisk in a call center setting? If so what what is your experience with it? Is that software really free for asterisk users? Or is there a better option out there? Thanks,

Hi Everyone, There are a few things I like in OrderlyStats, specially some graph presentations and the fact that if agent puts someone on HOLD or PAUSE it shows fine. 1 -But I see a lot of similarities in pricing, descriptions, wording on both sites. Were these same projects forked out? or is it still both owned by the same company? 2- What are the main differences between the two? - so that we

Hello Adam, Thank you so much for taking the time to write to me. I can understand your concerns; let me see if I can address them. >Matt, > >Sourceforge.net is exclusively for hosting software whose licensing >terms meet the OSI's definition of Open Source: > > http://opensource.org/docs/definition.php > >Your licensing terms include the following, which is

Hi all, Someone know how can I check for available members on a queue Before I queue the call, so I can do something else with it? Note that is not the case for joinempty Thanks, Gabriel Ortiz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090718/462b725b/attachment.htm

My abandon count seems to increment by two whenever there's an abandon... has anyone else noticed this? I'm on (more or less) the latest CVS... probably about a month old. I can't find anything in bugs.digium.com about it. Thanks, Chad

Hi all, After * starts the command "queue show" would not show any of the realtime queues, but just the ones that are in the queues.conf file. In this state de AMI would not send any "QueueMemberStatus" for that queues until a call is received by that realtime queue. Anyone knows any whay to load this information in *'s memory without the need of the queue receiving a

Hello everyone, I've just upgraded from 1.6.1.9 to 1.6.2.13. I have noticed that (after the upgrade) abandoned calls within the Queue produce no cdr at all. I am using unanswered = no (the default) in cdr.conf. The call shows, as expected, in the queue_log as ABANDON The dialplan is: Ringing(); Answer(); // need to answer or no music! goto

Hi, A have an asterisk connected to a legacy PBX trought an E1 and to the PSTN trought another E1. When the legacy user dial to the PSTN the call pass trought Asterisk. All works OK, the only problem is the delay on the Asterisk server when it receives the digits from the 1st E1 link. It will only make the call when the digit timeout expires. Is there a way to make something like

Hi I'm using Ast 1.0.3, and managing a Queue for ACD. Our Callcenter supervisor uses Flash Operator Panel to see status of the Queue, and logged in Agents. All calls that stay to long in the Queue gets into a Voicemail, in order to have customer leave a message. 1. I see the Queue has a Completed and Abandoned counter. I have not found a exact definition of these, besides the obvius ones.

Hi all, When I have 2 masks that would like to execute the same logic, there is the way to use the Goto (or any other) command without changing the ${EXTEN}? Eg. DID range is 1200-1349 -> call Macro(disca), what mask to use? (I just got it with 2 masks, but I didn't wanted to duplicate the dialplan for both) [test] exten => _12XX,1,Set(DIR=3) exten =>

Hi all, I'm trying to use the CDR() function on the "Variable" header of the Originate AMI action, but it isn't working. Anyone knows anything about this problem? asterisk 1.4.26 Thanks, Gabriel Ortiz -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello Jason, I've just come across your post to the Asterisk-Users group regarding OrderlyQ (from a web search - sorry it's taken so long). You wrote: > What experience can be shared about installing and running the OrderlyQ application? I have a bunch of > queues set up and want to start adding some additional apps and this one looked promising but I have > very

Hello all, >>From your system command line (not asterisk), type 'mpg123' and tell >> us what version of mpg123 you're running. >> >> If its not v0.59r or v0.59q, then get one of those installed. >> (Lots of notes say v0.59r only, however I've been using v0.59q >>. on RHv9 and Fedora 3 boxes with no problems.) Andrew wrote: > FWIW, I have

Hi all, I need to know the SIP response code from within the dial plan, someone could point me on how to? Gabriel Ortiz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090506/2e62bfda/attachment.htm

Greetings, list! My boss would like some statistics on how many calls are answered out of specific queues during a given time period, and I'm not sure how exactly to obtain those stats. Here's how our queue system works. 1) Call comes in and enters our 'ring' queue where the phones ring for 15 seconds (caller hears the standard ring tone). 2) After 15 seconds, the caller

I went to run my queue_log parser so that I could send out a monthly report to one of my customers, and I noticed that every valid call complete action (COMPLETEAGENT, COMPLETECALLER) is followed by an ABANDON: Here is a complete-caller: 1123325015|1123325011.2|mainq|NONE|ENTERQUEUE||00110102102 1123325020|1123325011.2|mainq|Agent/21|CONNECT|5

What experience can be shared about installing and running the OrderlyQ application? I have a bunch of queues set up and want to start adding some additional apps and this one looked promising but I have very little java experience and it doesn't seem to be running properly. Jason Kawakami -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, Does anyone uses astDB for a large amount of data, in special for implementing black lists with millions of numbers (i'd like about 2 or 3 million)? That would be held in memory right? Is this (memory consumption) the only problem I could face? Att. Gabriel -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I've tried search this problem on the list... no luck... The case is: without externip/localnet config on sip.conf [general] my SIP trunk works, but with no audio NAT problem (asterisk sends the private 192 address to the outside...) when I configure externip/localnet correctly my SIP trunk simply disappear! Checking the signalling with tcpdump shows me that Im sending the