provided by Star2Billing S.L.

OK, First of all i would like to thank Areski and all other Active members of this forum for thier continiues help and support.

This week i have installed Trixbox 2.2.4 with A2B 1.3.1 , every think was good except the CDR Report in the Trixbox admin sies every SIP-Friend call i place as an answered call,even if it is not answered.However In A2B Call Report everything is correct, if i answer the call it shows the correct call duration and cost, and if don't answer it says CANCEL in the TC field.

If i place a call from Trixbox Extention Everything is OK,at least in the trixbox CDR report. I first thought it has something to do with the known problem of some asterisk 1.2 versions, so i upgraded 1.2.4 , then 1.4.10 , then 1.4.17 , with the same result.

I then thought it is configuration problem and that asterisk answers the call too early, i changed the answer_call = in a2billing.conf to NO ,with the same result.

The problem is very clear in the log, but i just don't know why it behaves like that.

Jan 22 11:13:55 DEBUG[3746] chan_sip.c: Setting NAT on RTP to 0
Jan 22 11:13:55 DEBUG[3746] chan_sip.c: Outgoing Call for 00xxxxxxx
Jan 22 11:13:55 DEBUG[3746] channel.c: Driver for channel 'SIP/5079623989-0a0ef540' does not support indication 3, emulating it
Jan 22 11:13:55 DEBUG[3101] chan_sip.c: Acked pending invite 102
Jan 22 11:13:55 DEBUG[3101] chan_sip.c: Stopping retransmission on '[email protected]' of Request 102: Match Found
Jan 22 11:13:55 DEBUG[3101] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[email protected]' Request 103: Found
Jan 22 11:13:56 DEBUG[3101] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[email protected]' Request 103: Found
Jan 22 11:13:57 DEBUG[3101] chan_sip.c: Auto destroying call 'Yjg0MWJiMjZhOGU1M2RhODQ5ODIxOThkMmU5ZTk4Zjg.'
Jan 22 11:14:00 DEBUG[3485] manager.c: Manager received command 'Command'
Jan 22 11:14:00 DEBUG[3485] manager.c: Manager received command 'Command'
Jan 22 11:14:00 DEBUG[3485] manager.c: Manager received command 'Command'
Jan 22 11:14:09 DEBUG[3746] chan_sip.c: update_call_counter(00xxxxxx) - decrement call limit counter
Jan 22 11:14:09 DEBUG[3746] chan_sip.c: Acked pending invite 103
Jan 22 11:14:09 DEBUG[3746] chan_sip.c: Stopping retransmission on '[email protected]' of Request 103: Match Found
Jan 22 11:14:09 DEBUG[3746] app_dial.c: Exiting with DIALSTATUS=CANCEL.
Jan 22 11:14:09 VERBOSE[3746] logger.c: a2billing.php|1: file:Class.RateEngine.php - line:1157 - [USEDRATECARD=0]
Jan 22 11:14:09 VERBOSE[3746] logger.c: a2billing.php|1: file:Class.RateEngine.php - line:914 - [CC_asterisk_stop QUERY = INSERT INTO cc_call (uniqueid,sessionid,username,nasipaddress,starttime,sessiontime, calledstation, terminatecause, stoptime, calledrate, sessionbill, calledcountry, calledsub, destination, id_tariffgroup, id_tariffplan, id_ratecard, id_trunk, src, sipiax, buyrate, buycost, id_card_package_offer) VALUES ('1200996833.0', 'SIP/5079623989-0a0ef540', '5079623989', '', CURRENT_TIMESTAMP - INTERVAL 0 SECOND , '0', '00xxxxxxxx', 'CANCEL', now(), '0.025', '-0', '', '', 'NL', '1', '1', '1', '2', '151843420800390', '0', '0.009', '0', '0')]
Jan 22 11:14:09 VERBOSE[3746] logger.c: a2billing.php|1: file:Class.RateEngine.php - line:917 - [CC_asterisk_stop 1.1: SQL: DONE : result=1]
Jan 22 11:14:09 VERBOSE[3746] logger.c: a2billing.php|1: file:a2billing.php - line:310 - [a2billing account stop]
Jan 22 11:14:09 VERBOSE[3746] logger.c: a2billing.php|1: file:Class.A2Billing.php - line:650 - [CARD STATUS UPDATE : UPDATE cc_card SET inuse=inuse-1 WHERE username='5079623989']
Jan 22 11:14:09 DEBUG[3101] chan_sip.c: Stopping retransmission on '[email protected]' of Request 103: Match Found
Jan 22 11:14:09 DEBUG[3101] chan_sip.c: Stopping retransmission on '[email protected]' of Request 103: Match Not Found
Jan 22 11:14:09 DEBUG[3746] pbx.c: Extension 00xxxxxxx, priority 3 returned normally even though call was hung up
Jan 22 11:14:09 DEBUG[3746] chan_sip.c: update_call_counter(5079623989) - decrement call limit counter
Jan 22 11:14:09 DEBUG[3746] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
Jan 22 11:14:09 DEBUG[3746] 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) VALUES ('2008-01-22 11:13:53','151843420800390','151843420800390','00xxxxxxx','a2billing', 'SIP/5079623989-0a0ef540','SIP/mytrunk-0a10eeb8','Dial','SIP/mytrunk/00xxxxxxxxx|60|HRgrL(1199960000:61000:30000)',16,16,'ANSWERED',2,'5079623989')
Jan 22 11:14:11 DEBUG[3101] chan_sip.c: Stopping retransmission on '[email protected]' of Request 102: Match Found
Jan 22 11:14:15 DEBUG[3101] chan_sip.c: Stopping retransmission on '[email protected]' of Request 102: Match Found

the first time everything is good, it says exiting with dialstatus cancel, and it inserts the cancel status in the a2b call report, but it comes back and update again as an answered call and it insert that in the CDR Report of Trixbox.

and It only happens when i call from SIP Friend.

I still don't know if this is a bug or a mis configuration from me, but i wait what the experts say.

Thankz in advance!

I really want a sollution for this, can anybody help me plzzzzzzzzzzz!!

Are you sure you're not answering the call in the Asterisk dialplan before passing it to A2B?

thank you very much stavros, but is asterisk answering all calls by default,becouse i didnt change any thing in extensions.conf , and can you plz point to me to the wright direction, hier is my extensions.conf , i was 4 days busy with this problem zo plzz i really want to solve this today!!

thankz in advance!

; FreePBX
; Copyright (C) 2004 Coalescent Systems Inc (Canada)
; Copyright (C) 2006 Why Pay More 4 Less Pty Ltd (Australia)
; Released under the GNU GPL Licence version 2.

<stavros removed 50KBytes of useless information>

; include extension contexts generated from AMP
#include extensions_additional.conf

; Customizations to this dialplan should be made in extensions_custom.conf
; See extensions_custom.conf.sample for an example
#include extensions_custom.conf

Your default FreePBX extensions.conf didn't mention A2B at all, so I removed it as it wasted more bandwidth with each view than my first 3 computers combined had RAM.

I'm not going to lead you through this by the hand. At some point you're going to have to learn the basics of Asterisk. It may as well be now.

Stavros,

plz dont think that i am sitting back hier and just asking questions, as i said, i was busy with this problem for 4 days, making several installations and upgrades,but i just didnt solve the problem, and that is why i am asking!.

I have to say that i have treid with the follwing A2B config in extensions.conf under from-trunk but the result was same:

[a2billing]
; CallingCard application
exten => _X.,1,Answer
exten => _X.,2,Wait,2
exten => _X.,3,DeadAGI,a2billing.php
exten => _X.,4,Wait,2
exten => _X.,5,Hangup

I am not asking anybody to lead me and teach me hier but i am only asking that somebody just point me to the wright direction and tell me what i have to do and I will thank him a lot!!

thank you all!
[/quote]

Your earlier debug snippet showed that you were correctly launching A2B; adding presumably another [a2billing] context will not help, and could actively harm your installation.

My original question was "are you sure you're not answering the call in the Asterisk dialplan before passing the call to A2B". You've inadvertently answered that:Hint: Answer does just that.

Thats why i dont have that context in my extensions.conf , i just tryed once and i then removed it.

can this part of the log be the problem:

Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'from-did-direct' tries includes nonexistent context 'ext-findmefollow'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'from-pstn' tries includes nonexistent context 'from-pstn-custom'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'custom-meetme' tries includes nonexistent context 'ext-meetme'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'from-internal-additional' tries includes nonexistent context 'from-internal-additional-custom'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'app-blackhole' tries includes nonexistent context 'app-blackhole-custom'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'outrt-001-9_outside' tries includes nonexistent context 'outrt-001-9_outside-custom'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'outbound-allroutes' tries includes nonexistent context 'outbound-allroutes-custom'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'ext-local' tries includes nonexistent context 'ext-local-custom'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'ext-did-catchall' tries includes nonexistent context 'ext-did-catchall-custom'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'ext-did' tries includes nonexistent context 'ext-did-custom'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'ext-test' tries includes nonexistent context 'ext-test-custom'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'app-chanspy' tries includes nonexistent context 'app-chanspy-custom'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'app-zapbarge' tries includes nonexistent context 'app-zapbarge-custom'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'app-pickup' tries includes nonexistent context 'app-pickup-custom'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'app-userlogonoff' tries includes nonexistent context 'app-userlogonoff-custom'
Jan 23 00:14:36 WARNING[4812] pbx_ael.c: Unable to open '/etc/asterisk/extensions.ael': No such file or directory
Jan 23 00:14:36 WARNING[4812] pbx.c: Requested contexts didn't get merged
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'from-did-direct' tries includes nonexistent context 'ext-findmefollow'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'from-pstn' tries includes nonexistent context 'from-pstn-custom'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'custom-meetme' tries includes nonexistent context 'ext-meetme'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'from-internal-additional' tries includes nonexistent context 'from-internal-additional-custom'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'app-blackhole' tries includes nonexistent context 'app-blackhole-custom'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'outrt-001-9_outside' tries includes nonexistent context 'outrt-001-9_outside-custom'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'outbound-allroutes' tries includes nonexistent context 'outbound-allroutes-custom'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'ext-local' tries includes nonexistent context 'ext-local-custom'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'ext-did-catchall' tries includes nonexistent context 'ext-did-catchall-custom'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'ext-did' tries includes nonexistent context 'ext-did-custom'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'ext-test' tries includes nonexistent context 'ext-test-custom'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'app-chanspy' tries includes nonexistent context 'app-chanspy-custom'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'app-zapbarge' tries includes nonexistent context 'app-zapbarge-custom'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'app-pickup' tries includes nonexistent context 'app-pickup-custom'
Jan 23 00:14:36 WARNING[4812] pbx.c: Context 'app-userlogonoff' tries includes nonexistent context 'app-userlogonoff-custom'
Jan 23 00:14:36 VERBOSE[3136] logger.c: Reloading SIP


Plz tell me what to do and i will thank you very very much!

Those errors in your dialplan are certainly not going to be helping anything, but I'm pretty sure they're not responsible for answering the call in your [a2billing] context. I'm by no means an expert on FreePBX, but I find it hard to believe they ship the dialplan in that broken state.

Did the hint in my last post not suggest to you a simple change you could make yourself?
I'll state it more explicitly: you need to modify your [a2billing] context so that it no longer answers the call.
Digesting the information on this page should put you in a position where you can safely make this modification.