similar to: Asterisk 13. Writing call quality parameters to CDR. How?

because of problems you are facing i decided to go way with second table CREATE TABLE `cdr_extended` ( `id` int(11) unsigned NOT NULL AUTO_INCREMENT, `uniqueid` varchar(32) NOT NULL DEFAULT '', `callid` varchar(256) NOT NULL DEFAULT '' COMMENT 'sip call-id', `hangupcause` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL COMMENT 'info about

Dear Tariq; About elastix.org, this can be use with Asterisk or it is coming as a complete IP Telephony, Call Center, IVR and Reporting? Because, I do not need to install another IP Telephony on the server which already has asterisk which is an IP Telephony, this will cause a problem in the service (for example, when listening for SIP port of 5060).

Hello, I have a problem with some variables in 1.8.6.0. I set on extension the following lines: exten => h, n, Set (CDR (LLP) = $ {CHANNEL (rtpqos, audio, local_lostpackets)}) ; lost packets by local end ** exten => h, n, Set (CDR (PCR) = $ {CHANNEL (rtpqos, audio, remote_lostpackets)}) ; lost packets by remote end exten => h, n, Set (CDR (ljitt) = $ {CHANNEL (rtpqos,

Hi I have a small question on CDR Database: It's used by billing software no ? he have only one structure of data or they have multi structure with more information logged ? sample: cdr simple and cdr_extended thanks bye

Hello, How can I get peerip, recvip, from, uri, useragent, peername, t38passthrough variables in (within) my custom Asterisk application? I can't use chan_sip.c internal structures (such as sip_pvt) in my custom application, because there's no chan_sip.h and I can't include it into my application (maybe there's other way?). I can do like this: exten =>

Hi list! I'd like to save all information about calls (CDR) in a MySQL-Database. I created the DB and a user for Asterisk on a separate server, then I configured my cdr_mysql.conf so: [global] hostname=192.168.10.3 dbname=asterisk table=cdr password=MYSECRET user=asterisk port=3306 and my cdr.conf so: [general] enable=yes unanswered = yes safeshutdown=yes [mysql] usegmtime=no

I need some help understanding SIP dialog. Some actor is trying to access my server, but I can't figure out what he's trying to do ,or how. I'm getting a lot of these warnings. [May 17 10:08:08] WARNING[1532]: chan_sip.c:4068 retrans_pkt: Retransmission timeout reached on transmission _zIr9tDtBxeTVTY5F7z8kD7R.. for seqno 101 With SIP DEBUG I tracked the Call-ID to this INVITE :

Hello everyone, I am trying to activate Music On Hold using DB on Asterisk 13. It works fine but in order to use new Music On hold definitions I have to reload the moh module. - The following is my configuration in extconfig.conf - I added the following line: musiconhold.conf => mysql,asterisk,bit_ast_config - The following is the table in the database: mysql> select * from

06.09.2016 16:42, George Joseph ?????: > > > On Tue, Sep 6, 2016 at 7:32 AM, Dmitriy Serov <serov.d.p at gmail.com > <mailto:serov.d.p at gmail.com>> wrote: > > Hello. > > Several months server working on asterisk 13.7 and pjproject 2.5 > (installed separately). Once a day the server crashes or hangs and > is familiar sores that written

07.03.2015 0:24, Kevin Harwell ?????: > On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov <serov.d.p at gmail.com > <mailto:serov.d.p at gmail.com>> wrote: > > Hello. > > Asterisk 13.2. > I transfer configs from chan_sip to res_pjsip. > In chan_sip i have "match_auth_username=yes" and have nothing in > pjsip. > > I have a

-----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of jg Sent: Monday, July 06, 2015 4:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CDR in an MySQL-Database > Hi list! > > I'd like to save all information about calls (CDR) in a MySQL-Database.

Hello. Asterisk 13.2. I transfer configs from chan_sip to res_pjsip. In chan_sip i have "match_auth_username=yes" and have nothing in pjsip. I have a lot of endpoints and registrations on same SIP server. And it's problem in pjsip now. Is not it? I requesting to add new value for endpoint option identify_by. The value 'uri'. Simple config (cutted): [siptrunk]

Hi, I have the following code that operates when a channel is hung-up: [record-hangupcause]exten => 1,n,Set(CDR(hangupcause)=${HANGUPCAUSE})exten => s,n,Return() Before the dial a hangup handler is registered: Set(CHANNEL(hangup_handler_push)=record-hangupcause,s,1) The routine is called and the variables are being set, however not on the channel's CDR which made the call. I believe this

05.10.2015 23:24, Joshua Colp ?????: > On 15-10-05 05:22 PM, Dmitriy Serov wrote: >> Hello. Do I understand correctly that the current implementation >> res_pjsip does not support ZRTP? >> http://lists.digium.com/pipermail/asterisk-dev/2013-December/064401.html > > ZRTP is not supported in Asterisk itself. > >> Nothing has changed since 2013? P.S. I greatly

The call initiated from internal extension. I have made two test call: Successful call from device on res_pjsip via endpoint on chan_sip: http://pastebin.com/LWeDYstj Unsuccessful call from device on res_pjsip via endpoint on res_pjsip: http://pastebin.com/hepVb6Nu And ones again i don't see anything that would make asterisk send BYE. I would be grateful for any ideas. 11.02.2016 1:47,

The asterisk server has a permanent IP address, but the provider cannot ensure stable quality traffic for RTP. There is a desire to use an external server, the address of which shall be specified in the SDP, through which flowing media. I use asterisk 13.6 and res_pjsip. Prompt, please: 1. what software can be used on an external RTP proxy? 2. What settings need to be done in pjsip.conf to use

Hi, Anyone know how to set up tftp server for grandstream. I gues it should be somethink like <tftpserver-dir> <mac-address> firmware.bin config.txt Is this correct ? And how should the config-file look like. ? I had search sipphone.com but did'nt find anything. /HHA _________________________________________________________________ Rethink your

Hello. Do I understand correctly that the current implementation res_pjsip does not support ZRTP? http://lists.digium.com/pipermail/asterisk-dev/2013-December/064401.html Nothing has changed since 2013? P.S. I greatly regret that moved from chan_sip to res_pjsip. Previously used very much lacking, and much of the promise failed. Dmitriy Serov. -------------- next part -------------- An HTML

Good day. Asterisk 13.7.2, res_pjsip. There is a problem of loss of registration of several devices. This happens not on all devices, but problem devices a lot. Below is the log of registration of a contact of one device. Is suspect two things: 1. delete a contact after the contact is added. But, like, it's a feature of code that may already be fixed. 2. deleting a contact much earlier

Hello. After successful compilation 15.6.1 (bundled pjsip) and start asterisk i has error Symbol pjsip_tls_transport_start2 not found. /main/libasteriskpj.exports does not containg pjsip_tls_transport_start2 and pjsip_tls_transport_start. More: * All versions before (including 15.5) has not such error on this computer (ubuntu 18.04). * with 15.6.0, 15.6.1 has error on this computer