similar to: COMPLETEAGENT vs. COMPLETECALLER

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

Hei! Here's my problem. I have an Asterisk with SS7 and SIP trunks. Asterisk version is 1.6. I'm setting up a custom CDR fields and I was wondering is there a way to know who initiated a hangup? Asterisk must be aware of that info somehow, cause in queue_log, that info is present (completecaller, completeagent) Is there a way to get that info on the regular SS7 to SIP (and vica versa)

I am trying to figure out which one of our agents is answering the calls. According to http://www.voip-info.org/wiki/view/Asterisk+log+queue_log the only time the queue_log puts the channel (agent) is during logoff & logon. There is the connect & completeagent message, but it doesn't show which channel (agent) answered the phone. I can't even figure it our cross referencing the

Hi, Recently we got a new feature request from our customer, they want a report to list the duration that agents putting customer on hold, they want to base on this to measure the agents performance. I cannot find any events in cdr, message logs, or manager interface, only when I enable sip debug, then I can see the ReInvite Event in the cli , some thing like the attached logs, is there any

Hello all! I've "Queue Log Analyzer" installed [http://www.micpc.com/qloganalyzer] to check Asterisk main stats. There's an option of "CALLS COMPLETED [ALL]" where I can see the completed calls that entered any of the queues. There, there's a column that displays the extention number of the Agent who received each call, for example, it displays: Agent/202 So

I am experiencing several entries in the queue_log with a duration of NULL at the COMPLETEAGENT or COMPLETECALLER event. Any idea how this can happen? Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Gesch?ftsf?hrer: Stefan

Under 1.4.5 and 1.4.6, I've seen a few instances where the channel name in queue log entries is replaced by a snippet of a manager event: --START-- 1183582823|1183582823.104763|queuename|SIP/XXXX|REMOVEMEMBER| 1183582828|1183582793.104744|queuename| Context: macro-dialout Extension: s Priority: 3 Application: GotoIf AppData: 0?blockclid Uniqueid: 1183582822.104759

Hello Team, Can you help me on this? I have attached here the queue logs of my asterisk. I've searching a parser for this. I do not know what are the meaning of that logs. Thank you so much. Your response is highly appreciated. Regards, Cristina -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I'm working on an Asterisk configuration for a call center, and they bill based on the time spent talking to an agent, but not for any time spent waiting in a queue. The CDR information contains the entire duration of the call as billable seconds, including time spent waiting in the queue. I would like the billable seconds to only include the time spent actually talking to an agent.

Has anyone observed a problem where using Local channels with AddQueueMember results in missing TRANSFER events? Right now I'm using straight SIP channels when I call AddQueueMember(). I'm contemplating moving to Local channels because the non-state-based wrapuptime blows when you have a channel in multiple queues (they can hang up and get a call immediately so long as it's from a

Hi, For some reason (outbound call tracking) I've got a few different outbound call process (using a macro for queuemetrics logging, or direct call) i wanted to factorise the routing process so i came up with something like the following. All in one it's working like expected, however every "ael reload" command trigger a lot of warning like that "application call

Hello again List! I wrote something to allow me to easily view the state of Asterisk SIP devices, ZAP channels and Agents through a WEB interface. The pages are AJAX enabled to watch events and automatically update the status buttons. It is an Asterisk Switchboard created solely from Asterisk Events. It does not poll the Asterisk Manager (although it does have functions to run manager

Hi, I am running a call center based on Asterisk and building some statistics based on the queue_log file. I have some doubts about it that maybe you could help (actually, maybe these doubts are suggestions for enhancements!): 1st Scenario - Agent receives the call, and puts it on parking for somebody else to pick it up. Parking # 7000 (for attender transfer)

1 of our customers records all phone calls and needs to be able to be played back via a searchable web app. I tried ARI but it is very limited. Anyone have any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070518/40621109/attachment.htm

Hello list, I don't know if anybody faced this issue, but I finally found a workaround. I am using an external program with an AMI connection to originate outbound calls to Local/ channels, and on the dialplan context I dial outside with the corresponding trunk according to the prefix of the dialed number. The problem here was CDR, more precisely the uniqueid. Reading these mail lists I

The Asterisk Development Team has announced the release of Asterisk 11.3.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

Does anyone know why i would get this RINGNOANSWER events in queue_log when clearly the agent is busy and call-waiting is disabled. 1309550595|1309550570.399965|2253|Local/05 at from-internal/n|CONNECT|2|1309550593.399966|0 1309550632|1309550533.399961|2253|Local/11 at from-internal/n|COMPLETECALLER|1|74|1 1309550663|1309550640.399969|2253|NONE|ENTERQUEUE||zzzzzzzzzz

The Asterisk Development Team has announced the release of Asterisk 11.3.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

Hi, Surely, I must be overlooking something. If I run the following SQL queries I don't get the same number of rows. Is this coherent? mysql> select * from queue_log where queuename = '4010' and FROM_UNIXTIME(time) between 20080308000000 and 20080313145900 group by callid; 357 rows in set (0.01 sec) mysql> select * from cdr where dst = 4010 and calldate between 20080308000000

When I run this command: ssh -1 -v -n -l root njdb1b /usr/bin/X11/xterm -sb -sl 1000 -T njdb1b -e find /etc An xterm appears on my screen, the contents of /etc scroll by and then ssh exits. When I invoke the same command with the -2 option, the xterm appears (and disappears) but ssh never exits. netstat says the (ssh) connection is still established but the xterm is no longer