Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[asterisk-users] Asterisk 13. Writing call quality parameters to CDR. How?


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
serov.d.p at gmail.com
Guest





PostPosted: Wed Mar 18, 2015 2:37 pm    Post subject: [asterisk-users] Asterisk 13. Writing call quality parameter Reply with quote

Hello.

Voice quality when calling - this is one of the most important in the PBX.
You need to record the quality parameters for each call to improve.

Because the overall quality of a call can only be determined upon
completion, I did it in the HangUp handler and wrote in custom fields of
CDR.
This worked well in asterisk 11.

In asterisk 13 I did not find a handler after the call, but before
finalizing the CDR.
I tried to call the AGI and there to update the CDR record by unique
identifiers. But faced with the fact that there are no needed record in
the table yet.
To write the data into a separate table and join them may be an option.
But do not want to resort to such a decision

How do you solve this problem?

Dmitriy Serov.

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
cervajs at fpf.slu.cz
Guest





PostPosted: Thu Mar 19, 2015 4:12 am    Post subject: [asterisk-users] Asterisk 13. Writing call quality parameter Reply with quote

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 hangup',
`peerip` varchar(15) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
`recvip` varchar(15) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
`from_u` varchar(30) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
`uri` varchar(30) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
`useragent` varchar(30) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT
NULL,
`codec1` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
`codec2` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
`llp` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL
COMMENT 'lost packets by local end',
`rlp` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL
COMMENT 'lost packets by remote end ',
`ljitt` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL
COMMENT 'the same for jitter ',
`rjitt` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL
COMMENT 'the same for jitter ',
PRIMARY KEY (`id`),
KEY `uniqueid` (`uniqueid`)
) ENGINE=InnoDB DEFAULT CHARSET=utf8;

in hangup handler or h exten i will use func_odbc
like
insert into cdr_extended (uniqueid,hangupcause,peerip,...) values
('${UNIQUEID}',...);


Dne 18.3.2015 v 20:37 Dmitriy Serov napsal(a):
Quote:
Hello.

Voice quality when calling - this is one of the most important in the
PBX.
You need to record the quality parameters for each call to improve.

Because the overall quality of a call can only be determined upon
completion, I did it in the HangUp handler and wrote in custom fields
of CDR.
This worked well in asterisk 11.

In asterisk 13 I did not find a handler after the call, but before
finalizing the CDR.
I tried to call the AGI and there to update the CDR record by unique
identifiers. But faced with the fact that there are no needed record
in the table yet.
To write the data into a separate table and join them may be an
option. But do not want to resort to such a decision

How do you solve this problem?

Dmitriy Serov.



--
---------------------------------------
Marek Cervenka
=======================================


--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
ethy.brito at inexo.co...
Guest





PostPosted: Wed Mar 25, 2015 2:37 pm    Post subject: [asterisk-users] Asterisk 13. Writing call quality parameter Reply with quote

On Thu, 19 Mar 2015 10:12:22 +0100
Marek Cervenka <cervajs@fpf.slu.cz> wrote:

Quote:
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 hangup',
`peerip` varchar(15) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
`recvip` varchar(15) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
`from_u` varchar(30) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
`uri` varchar(30) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
`useragent` varchar(30) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT
NULL,
`codec1` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
`codec2` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
`llp` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL
COMMENT 'lost packets by local end',
`rlp` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL
COMMENT 'lost packets by remote end ',
`ljitt` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL
COMMENT 'the same for jitter ',
`rjitt` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL
COMMENT 'the same for jitter ',
PRIMARY KEY (`id`),
KEY `uniqueid` (`uniqueid`)
) ENGINE=InnoDB DEFAULT CHARSET=utf8;

in hangup handler or h exten i will use func_odbc
like
insert into cdr_extended (uniqueid,hangupcause,peerip,...) values
('${UNIQUEID}',...);

Interesting approach.

But how to tell from a call going directly (directmedia) and a call with
asterisk in between??

In the last case, two bridged channels, how to collect the parameters from each "leg" in the "h" extension?

Cheers

Ethy


Quote:


Dne 18.3.2015 v 20:37 Dmitriy Serov napsal(a):
Quote:
Hello.

Voice quality when calling - this is one of the most important in the
PBX.
You need to record the quality parameters for each call to improve.

Because the overall quality of a call can only be determined upon
completion, I did it in the HangUp handler and wrote in custom fields
of CDR.
This worked well in asterisk 11.

In asterisk 13 I did not find a handler after the call, but before
finalizing the CDR.
I tried to call the AGI and there to update the CDR record by unique
identifiers. But faced with the fact that there are no needed record
in the table yet.
To write the data into a separate table and join them may be an
option. But do not want to resort to such a decision

How do you solve this problem?

Dmitriy Serov.



--
---------------------------------------
Marek Cervenka
=======================================


--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


--

Ethy H. Brito /"\
InterNexo Ltda. \ / CAMPANHA DA FITA ASCII - CONTRA MAIL HTML
+55 (12) 3797-6860 X ASCII RIBBON CAMPAIGN - AGAINST HTML MAIL
S.J.Campos - Brasil / \

PGP key: http://www.inexo.com.br/~ethy/0xC3F222A0.asc

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services