similar to: [Re: CDR Rewrite -- Questions to the users]

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, I want to get a histogram with the legend for my data. I drew a normal density curve and kernel density curve in the histogram, and I also label mean and median in the X axis. From the code, I got two legend: One shows "Normal Density" and "Kernel Density" and their corresponding lines, the other shows "Mean = value" and "Median = value" and their

Buenas noches; Además del proyecto que comenté antes y con el que sigo discutiendo, también me estoy peleando con otro... con el que también tropiezo. Necesito reordenar los datos presentados en dos columnas a filas y columnas. Los datos ideales serían algo así: Ques <- c(rep("Q1",3),rep("Q2",3),rep("Q3",3)) Info <- rep(c("aca", "ahi",

Hello! It's me again. I began a fairly large modification to my CDR proposal some weeks ago, and finally yesterday and this morning got enough accomplished to allow a commit and some peer review. Check the docs out via " svn co http://svn.digium.com/svn/asterisk/team/murf/RFCs " This is a directory; in it you will find: CDRfix2.rfc.doc CDRfix2.rfc.docx CDRfix2.rfc.pdf The docx

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 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 all, It 's dificult to explain; The system I need is an box option (based on *), that I would add to an existing PABX (ie: Nortel with 600 ext). I need two E1/T2 card to plug the system between Telco (FT) and PABX (Nortel)! One card for France Telecom Side (E1a) and one other to Nortel Side (E1b). --------- -------- ----------- Telco FT

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

Hi all, I am currently struggling to remove one of our Samba4 DC from the domain. Some time ago, adding a new Samba DC to our AD did not succeed and I had to demote the new server again. After removal, replication on one of the old/existing DCs got weird. /usr/local/samba/bin/samba-tool drs showrepl gives the following: Standardname-des-ersten-Standorts\dc02 DSA Options: 0x00000001 DSA object

Sounds like Comcast's manual for CALEA compliance was leaked. Pretty interesting read if you are curious: http://www.fas.org/blog/secrecy/ Direct link (PDF): http://www.fas.org/blog/secrecy/docs/handbook.pdf -- Kristian Kielhofner

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

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 =>

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

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

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,

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Dear Centos Community: Allow me to introduce myself. I am Robert {Bob} Lightfoot from Borden, Indiana USA. My wikilogin shows a username of RobertLightfoot although on IRC and Fedora I am known as BobLfoot. I was reading thru the material found at http://wiki.centos.org/Contribute#head-42b3d8e26400a106851a61aebe5c2cca54dd79e5 in preparing for

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 was just reading through Info Week while on a flight and happened upon an brief piece about a new VOIP security intiative worked up by a handful of the usual suspects; Alcatel, SMU, NIST, Symantec, etc. All of this begs the question of can't we get just do this as a user community? I understand that the Zultys phone, which I own several, support AES encryption of the RTP stream.