similar to: Setting CDR variables for all linked channels

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

the content of th character vector (of length 1) is as follows: a <- "something2 ....pat1 name1 pat2 something2....pat1 name2 pat2....pat1 name3 pat2 " I would like to extract the character bewteen pat1 and pat2. That's to say, I would like to get a vecter of c("name1", "name2","name3"). What I did is use strsplit() twise. But I wonder if there

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

Hi all. I've being reading posts from the list since yesterday and I feel this question was answered a lot time ago, but the list archives are a mess (yet). I hope some one is willing to help me out. I want to set up this: caller ----- PSTN ---- (SOMETHING1) ------ VoIP --------- (SOMETHING2) ---- PSTN I think this must be a very basic architecture, but I'm not sure wat SOMETHING1

Hello, In a couple of Asterisks, after type "sip show channels" we have a lot of these: IP_PEER dst_number something 00102/00103 unkn No (d) Rx: BYE IP_PEER dst_number2 something2 00102/00103 unkn No (d) Rx: BYE We are using ASterisk 1.2.x When I say "a lot" I mean more than 180, more than 230, etc. Is it normal? How we can remove it? Thank you very much, --

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 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

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

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

Hello, Does samba support the use of S4U? What do we need to configure in SAMBA or krb5 to support getting a ticket obtained by S4U. We are using 3.0.25 and krb5-1.4.1 We are getting the following error: decode_pac_data: Name in PAC [username@something1.something2.realmname] does not match principal name in ticket The ticket could be different than the PAC name because the

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

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

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

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:

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

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

http://bugzilla.mindrot.org/show_bug.cgi?id=1039 Summary: Incomplete application of HostKeyAlias in ssh Product: Portable OpenSSH Version: 4.0p1 Platform: All OS/Version: All Status: NEW Severity: normal Priority: P2 Component: ssh AssignedTo: bitbucket at mindrot.org ReportedBy: cdmclain

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 =>

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