similar to: Incoming calls not showing up in user specific CDRs?

Hi For whatever reason we've getting 2 or 3 CDR lines logged for each call, often in different formats: as1:~# grep test-89-1e2c /var/log/asterisk/cdr-csv/*.csv

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

Hello, I'm working on some dialplan rules to pull multiple users into a conference call. I have some fairly straightforward rules which start up a new MeetMe conference, allow escape with the * key to invite more users, then use a features.conf sequence to bring the new user into the conference with ChannelRedirect. The problem I'm running into is the time in the MeetMe conference

Hi a new small question ;=) We have two Asterisk, connected in IAX2. On the first, in dialplan, we have: exten => _XX.,1,Set(IAXVAR(ACCOUNTID)=${CDR(accountcode)}) we sent into the IAXVAR "ACCOUNTID" the accountcode. On the second, in dialplan, we have: exten => 18,2,AGI(Caller-ID.agi,${IAXVAR(ACCOUNTID)}) That's work, the second server get the variable. I

Hi all, I have about 10 SIP phones for different users defined in sip.conf, each with their own accountcode= entry. There is a global setting in sip.conf that states amaflags=documentation There are 3 IAX->PSTN gateways defined in iax.conf for outbound calls. These do not have an accountcode=, but do have amaflags=billing defined in each. The theory was that all calls should be logged, those

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

Please help with this strange issue. When "sip show peers" returns status "Unknown" the CDR does not include the accountcode even though the call is correctly processed. I'm using A2 Billing and it uses the accountcode to determine the authentication. Asterisk version 1.4.21.2 I'm calling from a Quintum device. I'm very puzzeled. Name/username Host

Skipped content of type multipart/alternative-------------- next part -------------- GRANT INSERT ON cdr.* TO root@localhost IDENTIFIED BY '8anana?!'; USE cdr; CREATE TABLE cdr ( uniqueid varchar(32) NOT NULL default '', userfield varchar(255) NOT NULL default '', accountcode varchar(20) NOT NULL default '', src varchar(80) NOT NULL

Test ------Mensaje original------ De: tracinet Remitente:asterisk-users-bounces at lists.digium.com Para:Asterisk Users Mailing List - Non-Commercial Discussion Responder a:Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP Enviado: 6 Mar, 2009 5:55 PM Basically, Server 1 is the main customer PBX where we have multiple

Hello, We have finally upgraded to Asterisk 1.4, however we've run into two issues that weren't occurring before the upgrade. Issue #1: We're an outgoing call center and need to record all calls. We use the uniqueid field in the CDR to match with the recording, which we labeled with {UNIQUEID} in MixMonitor. For some reason, the uniqueid is not correct in the CDR. Here is the

Hello, I have a TDM400P with 2x2 configuration of FXOs and FXSs. I set a test extension of '444' to dial out a specific zap trunk and call a local #. First time I call out to '444' everything works fine. If I hang up the call, and within 10 seconds dial the same number again, I get a fast busy. Seems it isn't letting go of the trunk or something, and I don't have a

Both of our production asterisk servers are dumping core when making writes to our cdr tables. Here is a backtrace of the problems we are having: #0 0x00447b1f in tdserror (tds_ctx=0x1, tds=0xb7938c90, msgno=20004, errnum=9) at util.c:347 347 if (tds_ctx && tds_ctx->err_handler) { (gdb) bt #0 0x00447b1f in tdserror (tds_ctx=0x1, tds=0xb7938c90, msgno=20004, errnum=9) at

Hi Folks, I figured out how to get the AMAFlags set correctly! It was not documented anywhere I could find, and I grepped through the source code trying to track through the different function calls etc. all to no avail. But... if you note in the sample.call file, they show how to do Set(CDR(userfield|r)=whatever. I tried adding this "|r" piece into amaflags, like so:

For the information of others (and Google) this is the table structure I used to get cdr_mysql working: CREATE TABLE cdr ( calldate varchar(255) NOT NULL default '', clid varchar(255) NOT NULL default '', src varchar(255) NOT NULL default '', dst varchar(255) NOT NULL default '', dcontext varchar(255) NOT NULL default '', channel varchar(255)

Hello I just settup a realtime mysql table for sip_peers. All peers (friends) is autenticateing but when i want to initiate a call between them i got the following error. Someone have some ideea? Thank you. ---<Cut Here>--- pbx*CLI>console dial 1014 == Console is full duplex -- Executing [1014@default:1] Dial("OSS/dsp", "SIP/1014|40|t") in new stack

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

On Sat, 25 Apr 2015 17:11:34 +0200 jg <webaccounts173 at jgoettgens.de> wrote: > > > Hi All > > > > I have dozens of these messages on CLI complaining about database > > connection and error writing CDR to disk. > > > > The curious thing is I can find them all inside the database. > > I "selected" them using uniqueid and manually

Hi, I'm currently running CVS-HEAD 2005-09-03 I do plan to upgrade to the newest version, but need to do some testing with it first. In the mean time... does anyone know what these messages below are about? I've never seen it before, but when it happened it locked Asterisk up pretty good. Feb 10 10:16:51 DEBUG[14917] chan_zap.c: Echo cancellation already on Feb 10 10:16:57

Hello All Anybody had used ooH323 for asterisk i using ooH323-0.7.2 and asterisk CVS may 2005. OpenH323 1.17.1 and pwlib 1.9.0 and GNUGK 2.0.2 audio is very good, better than SIP and IAX, but i have problem. how to router call from openh323 to outside PSTN. my h323.conf setting ; Objective System's H323 Configuration example for tvcti ; ooh323c driver configuration ; ; [general]

Not being an asterisk expert, but having been around the block once or twice when it comes to data and the like, I have made some observations based on the examples given on voip-info.org Sip configs. it appears there is an adjustment to be made in the sip_buddies example table: >>> name Although set to 30 characters, I don't see where it is limited in the text file. In theory,