similar to: CEL events

Hi, I've already post this to the forum three days ago, sorry if it's sounds like a crosspost, but I've got no replies, so I'm trying other channels :) This is the link to the forum post if someone prefer to reply here: http://forums.asterisk.org/viewtopic.php?f=1&t=86985 I'm using Asterisk 1.8.20.0 (the freepbx build) with CEL logging activated. I'm using CEL because

On Wed, Jul 22, 2020 at 12:44 PM Administrator <admin at tootai.net> wrote: > No one on this ? > > Le 10/07/2020 à 18:06, Administrator a écrit : > > Hi, > > > > On Asterisk 16.11.1 when enabling cel I get error with BRIDGE_START > > and BRIDGE_END events > > > > zone-s*CLI> module reload cel > > The module 'cel' reported a

Hi, On Asterisk 16.11.1 when enabling cel I get error with BRIDGE_START and BRIDGE_END events zone-s*CLI> module reload cel The module 'cel' reported a reload failure     -- Reloading module 'cel' (CEL Engine) [2020-07-10 17:57:01] ERROR[16163]: cel.c:428 ast_cel_str_to_event_type: Unknown event name 'BRIDGE_START' [2020-07-10 17:57:01] ERROR[16163]:

No one on this ? Le 10/07/2020 à 18:06, Administrator a écrit : > Hi, > > On Asterisk 16.11.1 when enabling cel I get error with BRIDGE_START > and BRIDGE_END events > > zone-s*CLI> module reload cel > The module 'cel' reported a reload failure >     -- Reloading module 'cel' (CEL Engine) > [2020-07-10 17:57:01] ERROR[16163]: cel.c:428 >

-----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,

Hi, I have been experimenting with CEL in a trunk version of asterisk for some time and have upgraded my test machine to 1.8.0 today. Made a few calls and it looks like the eventtype field is missing in the CEL insert query when using ODBC. I see the following errors on the console: [Oct 22 21:46:09] WARNING[952]: res_odbc.c:634 ast_odbc_prepare_and_execute: SQL Execute returned an error -1:

Dear friends, Need to know filenames of conference recordings in Asterisk 11. Besides directory scanning the recordings could use CEL: Filter MySQL rows with eventtype equal CHAN_START and channame like ConfBridgeRecorder and then get the eventtime field and convert to timestamp to complete filename(s). Would you suggest any other approaches? Thanks. -------------- next part -------------- An

Dear friends, In Asterisk 11.7.0, is it possible to receive CEL APP_START and APP_END events from Park application? Queue and Dial apps are generation these events, but Park no. It doesn't seem to make difference when configured. cel.conf ... apps = dial,queue,confbridge,park events = ALL ... localhost*CLI> cel show status ... CEL Tracking Application: queue CEL Tracking Application:

An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130630/18883d29/attachment.htm> -------------- next part -------------- A non-text attachment was scrubbed... Name: 1371568201.486360-80.csv.zip Type: application/zip Size: 1674 bytes Desc: not available URL:

my program given below can some one make it presentable. I trying to simulate survival data and calculate the power. I think i could have done better. s=10 number=0 count1=0;count2=0;count3=0;count4=0;count5=0;count6=0;count7=0;count8=0; count9=0; count11=0;count22=0;count33=0;count44=0;count55=0;count66=0;count77=0; count88=0;count99=0; while(s!=0){ n=100 n1=n/2 n2=n/4

I was trying to recreate this kind of timeline plot: http://www.vertex42.com/ExcelArticles/create-a-timeline.html As you can see in their excel example, the events are nicely placed out on both sides of the timeline axis. AFAIK there is no function to do this nicely in R-project. Furthermore, graphics and lattice packages are unable to draw the x-axis in the middle of the plot. (datapoints

Hi all, Just thought I'd update this thread in case anyone else is Googling trying to find out how to do this... I found the solution to my problem to be to use the IAXVAR() function to pass the accountcode between the Asterisk boxen and update the CHANNEL(accountcode) with that variable. Thanks to Richard @ Digium for the reply that clarified my misunderstanding. Calum On Wed, 2018-10-03

Hi asterisk-users, We have recently moved to the 13.x branch of Asterisk from 11.x, and we're trying to correlate CDR records from multiple-legs for billing purposes. As part of this change we have added 'linkedid' to our CDR table schema in an attempt to join the multiple records into one billable record. The call path can be simplified as (transport types in brackets): SIP

I am trying to get cdr via odbc to work on Asterisk 13.7.2 but I keep getting this error: [Feb 9 16:21:43] WARNING[2088]: cdr_odbc.c:160 execute_cb: cdr_odbc: Error in ExecDirect: -1, query is: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield,peeraccount,linkedid,sequence) VALUES ({ts

Benny-- Thanks for the response! I've inserted comments in the following: PS. Pardon the HTML format; my email editor splits lines at an unadjustably small number of columns, but in HTML, no line length limits, and better looking examples! On Tue, 2009-01-13 at 14:16 +0100, Benny Amorsen wrote: > Steve Murphy <murf at digium.com> writes: > > > Which of the two would

Hi! my asterisk-12.2.0 with pjsip-2.2.0 does not translate codecs any more. I tried every combination. silent on both sides. I compiled pjsip with no resample in pjsip. ./configure --prefix=/usr --enable-shared --disable-sound*--disable-resample* --disable-video --disable-opencore-amr is there a way to force asterisk back to do the codec translation? Attachment: sip show channel of the

our * crashed twice in a month with segmentation fault & a core dump. here's the stack trace: #0 0xb7e11965 in mallopt () from /lib/tls/libc.so.6 #1 0xb7e10c43 in malloc () from /lib/tls/libc.so.6 #2 0xb7e17090 in strdup () from /lib/tls/libc.so.6 #3 0x08057ada in ast_verbose (fmt=0x0) at logger.c:879 #4 0xb7b29009 in moh_files_release (chan=0x9455ca0, data=0xb657be48) at

I am configuring a test Asterisk server (1.8.9.2) to practice setting a single codec globally, to avoid transcoding as much as possible. Since all of my recordings are in gsm format, I am trying to make the SIP clients use gsm everywhere. I am using Ekiga on Fedora 16 x86_64 for my tests. [root at elx2 asterisk]# cat /etc/asterisk/sip_general_additional.conf

Greyman-- I'm taking this discussion to the list. Folks, what we are talking about here, is me trying to get a grasp around Greyman's (Aaron's) request for a bare-bones CDR generation that describes just total connect time for channels, stripping out all the details. Who cares about xfer, park, hold, etc.? So in the following is our discussion about what *should* be there, and in

Hi, I am using Asterisk 11.2.0. Channel Event Logging (CEL) ist activated and running. CEL entries are logged into an mysql database. So far so good. I want to do some extra cel logging and try the following via an AGI-Script: EXEC CELGenUserEvent test In the asterisk logfile I can see the following: -- AGI Script Executing Application: (CELGenUserEvent) Options: (test) (no errors or warnings)