search for: linkedids

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

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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

Hello! Most are probably bored seeing another letter about this, but I've put in a fair amount work on a spec for rewriting the CDR system in Asterisk, and I have some questions: First, please look at what I've written so far: svn co http://svn.digium.com/svn/asterisk/team/murf/RFCs and look at the file "CDRfix2.rfc.txt" in the RFCs dir. The spec SIGNIFICANTLY alters the way

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 Matt, I am trying to replicate this particular problem. We are seeing more frequently where the Event: AsyncAGIExec is never being sent. The two scenarios I have seen in tests yesterday and today... We sendl an AMI action. For example, play a short file or hangup. AMI Events will indicate it did the work, but we never receive the Event: AsyncAGIExec with a result at all. Asterisk debug

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

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

After an upgrade to Asterisk 12, I'm "collecting" channels. When I enter and then exit a conference room, I see: -- <CBAnn/207-0000067f;1> Playing 'confbridge-leave.slin' (language 'en') -- Channel CBAnn/207-0000067f;2 joined 'softmix' base-bridge <5edb1920-3774-4ba3-8c4d-23e8fd04519c> -- Channel CBAnn/207-0000067f;2 left

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

I'm having an issue with CDR. Basically, I expect to have all "legs" of a call having the same linkedid and differing only by the sequence value. That does happen, but I'm getting null dst values after doing an attended transfer. I'm not sure if this is a bug or I'm doing something wrong. I'm running Asterisk 13.2.0. Here's the console log, step by step: First,

I understand how to control the first local channel, but an having trouble getting the second local channel to enter stasis. I setup have the following extensions.conf to handle 1000 (basically had it setup so if first stasis not there try second, but believe second channel never processes the dial plan so even if second line was hello-world2 it would not matter. [mycontext] exten =>

We are communicating with Asterisk via AMI. Running Asterisk version 13.18.5 on an Ubuntu box. If you look at the event response, the Result field is filled with random characters. I'm not sure what to do because that is a completely random result. It makes no sense. We send the following command to asterisk via AMI Action: AGI ActionID: C44415 Channel: SIP/192.168.40.105-00001338

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

Dear list. This is probably a complex subject but is that right to consider: a) each distinct linkedid field value in a mysql CEL table as a unique call? b) the duration of a call as the period (eventtime fields) between BRIDGE_END and BRIDGE_START events of the same linkedid sequence? (not considering transfers) Just a start ... Tks. -------------- next part -------------- An HTML attachment

Hi, when one does "Set(CDR(var)=value)" in dialplan, the value is only set for one record in the cdr table, but not the linked ones (the ones with the same linkedid). E.g. if you do something like same => n, Set(CDR(var)=value) same => n,Dial(Local/something&Local/something2) like only the original CDR record with have "var" set to "value", but the

Hi list! Running Asterisk 20.0.0 on CentOS 7, logging CDRs using cdr_adaptive_odbc to mariadb-server-5.5.68 (via mariadb-connector-odbc-3.1.7-ga-rhel7) Using chan_sip. I'm facing the problem when there is a sudden spike of calls, some of the calls that are being made during those spikes hang forever basically. This looks like this: [root at voip]# asterisk -rx 'core show channels

Hi, all I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded from asterisk.org). We named it "Asterisk11". I want to generate a call file to /var/spool/asterisk/outgoing. This call will dial out to Local Channel and return to some Extens. Then Asterisk11 will generate a CDR records to MySQL's cdr table(in database "mydatabase") via cdr_adaptive_odbc.

I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal: http://svn.digium.com/svn/asterisk/team/murf/RFCs. After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation

This is just a note that the fixes in the CDRfix4 and CDRfix6 branches are getting closer to being merged into 1.4, trunk, and 1.6.x. If CDR's are important to you, and you ignore this notice, then you deserve what you get! These branches address various long-standing bugs, most of which are regressions from 1.2. It is hoped that these fixes will solve most of the problems introduced by the