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 Post subject: billing never stop after connection break
PostPosted: Tue Aug 15, 2006 9:31 am 
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Joined: Tue Aug 15, 2006 9:07 am
Posts: 13
Dear forum people,
This morning I made international call by x-lite application and after call was established and talk about few minutes I loose Internet connectivity.

Nothing strange except that a2billing didn't register this and continue spending my credit to the zero $.

I'm not sure is this a Bug or there is an option that could detect connectivity problems.


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 Post subject:
PostPosted: Tue Aug 15, 2006 10:47 am 
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Joined: Sun Mar 12, 2006 2:49 pm
Posts: 954
Location: Barcelona
this is sounds more as an asterisk issue.
Once we are on the Dial command the hand is give to asterisk.
Which version do u run ?

Rgds, Areski


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 Post subject:
PostPosted: Tue Aug 15, 2006 11:16 am 
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Joined: Tue Aug 15, 2006 9:07 am
Posts: 13
Currently I'm on Asterisk@Home :

Asterisk 1.2.7.1 built by root @ asterisk1.local on a i686 running Linux on 2006-04-27 07:35:57 UTC
Verbosity is at least 3


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 Post subject:
PostPosted: Tue Aug 15, 2006 11:46 am 
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Joined: Tue Aug 15, 2006 9:07 am
Posts: 13
Here are details from /var/log/asterisk/full

Aug 14 22:08:34 VERBOSE[17695] logger.c: -- Executing Answer("SIP/3436606653-3b28", "") in new stack
Aug 14 22:08:34 VERBOSE[17695] logger.c: -- Executing Wait("SIP/3436606653-3b28", "2") in new stack
Aug 14 22:08:36 VERBOSE[17695] logger.c: -- Executing DeadAGI("SIP/3436606653-3b28", "a2billing.php") in new stack
Aug 14 22:08:37 VERBOSE[17695] logger.c: a2billing.php: [agi_channel] => SIP/3436606653-3b28
Aug 14 22:08:37 VERBOSE[17695] logger.c: a2billing.php: 3436606653 ; SIP/3436606653-3b28 ; 1155586114.1287 ; 3436606653 ; 00381XXXXXXXXX
Aug 14 22:08:37 VERBOSE[17695] logger.c: -- SIP/asterisk2pstn-2a17 is making progress passing it to SIP/3436606653-3b28
Aug 14 22:08:58 VERBOSE[17695] logger.c: -- SIP/asterisk2pstn-2a17 is making progress passing it to SIP/3436606653-3b28
Aug 14 22:09:03 VERBOSE[17695] logger.c: -- SIP/asterisk2pstn-2a17 is making progress passing it to SIP/3436606653-3b28
Aug 14 22:09:08 VERBOSE[17695] logger.c: -- SIP/asterisk2pstn-2a17 answered SIP/3436606653-3b28
Aug 15 05:07:12 DEBUG[17695] channel.c: Bridge stops bridging channels SIP/3436606653-3b28 and SIP/asterisk2pstn-2a17
Aug 15 05:07:12 VERBOSE[17695] logger.c: a2billing.php: INSERT INTO call (uniqueid,sessionid,username,nasipaddress,starttime,sessiontime, calledstation, terminatecause, stoptime, calledrate, sessionbill, calledcountry, calledsub, destination, id_tariffgroup, id_tariffplan, id_ratecard, id_trunk, src) VALUES ('1155586114.1287', 'SIP/3436606653-3b28', '3436606653', '', CURRENT_TIMESTAMP - INTERVAL 25084 SECOND , '25084', '00381XXXXXXXXX', 'ANSWER', now(), '0.25', '104.51666666667', '', '', 'serbia and montenegro cellular', '6', '10', '2475', '1', '3436606653' )
Aug 15 05:07:14 DEBUG[17695] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid) VALUES ('2006-08-14 22:08:34','\"3436606653\" <3436606653>','3436606653','00381XXXXXXXXX','callingcard', 'SIP/3436606653-3b28','SIP/asterisk2pstn-2a17','Dial','SIP/asterisk2pstn/00381XXXXXXXXX|30|HL(25055000:61000:30000)',25120,25120,'ANSWERED',2,'3436606653','1155586114.1287')


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 Post subject:
PostPosted: Tue Aug 15, 2006 2:14 pm 
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Joined: Sun Mar 12, 2006 2:49 pm
Posts: 954
Location: Barcelona
Well still sounds as Asterisk didnt notice to res_agi the call was disconnected.
honestly I never heard about such issues.
please give a try with asterisk 1.2.10

Rgds, Areski


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 Post subject:
PostPosted: Tue Aug 15, 2006 2:42 pm 
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Joined: Sun Jun 25, 2006 9:13 am
Posts: 183
Location: Germany
Hello,

Stop asterisk and see, some time call still open.


Mohan


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 Post subject:
PostPosted: Tue Aug 15, 2006 3:03 pm 
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Joined: Sun Mar 12, 2006 2:49 pm
Posts: 954
Location: Barcelona
stop asterisk or cut connection ?
and what happens ?


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 Post subject:
PostPosted: Tue Aug 15, 2006 3:23 pm 
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Joined: Sun Jun 25, 2006 9:13 am
Posts: 183
Location: Germany
Hello areski,


i had same problem , call
disconnected user site. but not in server ,2 days later when i have restated the asterisk i have found that call got disconected. Asterisk issue,

i here is cdr
2006-07-24 15:55:54 492002126526 983116687560 iran 2926:13 6011652054 ANSWER STANDARD 0.06 163.868 EUR


Note: Now i have set in sip.conf, problem never repeated .
rtptimeout=60
rtpholdtimeout=60

Mohan


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 Post subject:
PostPosted: Wed Aug 16, 2006 6:54 am 
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Joined: Tue Aug 15, 2006 9:07 am
Posts: 13
Areski, I repeat that this problem is hapening when I establish a call by x-lite and at that moment turn out ethernet cable from pc. But this problem is solved by Monah76.

rtptimeout=60
rtpholdtimeout=60

Thanks Monah76, this helps me to!

Aug 16 08:48:05 DEBUG[7111] channel.c: Didn't get a frame from channel: SIP/1233434513-cbbcAug 16 08:48:05 DEBUG[7111] channel.c: Bridge stops bridging channels SIP/1233434513-cbbc and SIP/asterisk2pstn-137d


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