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 Post subject: Loop issues?
PostPosted: Sat Feb 28, 2009 6:38 am 
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Joined: Mon Aug 18, 2008 7:02 pm
Posts: 15
Hi Guys,

I need some help as I think this is not asterisk issues instead, it might be a2billing

Asterisk: 1.4.22-3
A2Billing: 1.3.3

Quote:
[Feb 28 14:22:33] -- Limit Data for this call:
[Feb 28 14:22:33] > timelimit = 2147483647
[Feb 28 14:22:33] > play_warning = 61000
[Feb 28 14:22:33] > play_to_caller = yes
[Feb 28 14:22:33] > play_to_callee = no
[Feb 28 14:22:33] > warning_freq = 30000
[Feb 28 14:22:33] > start_sound = (null)
[Feb 28 14:22:33] > warning_sound = timeleft
[Feb 28 14:22:33] > end_sound = (null)
[Feb 28 14:22:33] -- Called sipprovider/65mynumber
[Feb 28 14:22:44] -- SIP/sipprovider-0954ba10 is making progress passing it to IAX2/trixbox-th-peer-6526
[Feb 28 14:22:47] -- Got SIP response 486 "Busy Here" back from sipprovider-IP
[Feb 28 14:22:47] -- SIP/sipprovider-0954ba10 is busy
[Feb 28 14:22:47] == Everyone is busy/congested at this time (1:1/0/0)
[Feb 28 14:22:47] -- Playing 'prepaid-isbusy' (escape_digits=#) (sample_offset 0)
[Feb 28 14:22:50] a2billing.php|1: file:Class.RateEngine.php - line:1155 - [USEDRATECARD=0]
[Feb 28 14:22:50] a2billing.php|1: file:Class.RateEngine.php - line:899 - [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 ('1235802141.113', 'IAX2/trixbox-th-peer-6526', '3451614166', '', CURRENT_TIMESTAMP - INTERVAL 0 SECOND , '0', '65mynumber', 'BUSY', now(), '0.0708', '-0', '', '', 'singapore (mobile)', '1', '1', '12999', '15', '66-calledfrom', '0', '0', '0', '0')]
[Feb 28 14:22:50] a2billing.php|1: file:Class.RateEngine.php - line:902 - [CC_asterisk_stop 1.1: SQL: DONE : result=1]
[Feb 28 14:22:50] a2billing.php|1: file:a2billing.php - line:312 - [a2billing account stop]
[Feb 28 14:22:50] a2billing.php|1: file:a2billing.php - line:985 - [CALLBACK 1ST LEG]:[INFO FOR THE 1ST LEG - callback_username=
[Feb 28 14:22:50] a2billing.php|1: file:Class.A2Billing.php - line:1792 - Requesting DTMF, CARDNUMBER_LENGTH_MAX 15
[Feb 28 14:22:50] -- <IAX2/trixbox-th-peer-6526> Playing 'prepaid-enter-pin-number' (language 'en')


It seems the call went into a loop after the call is detected as Busy/Congestion or even hangup from user side.

I try with a few provider and all are getting the same symptom

thanks


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 Post subject:
PostPosted: Sat Feb 28, 2009 11:24 am 
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Posts: 2890
Location: Devon, UK
Have you set 'number_try = 1' in a2billing.conf?


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 Post subject:
PostPosted: Sun Mar 01, 2009 1:58 am 
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Posts: 15
Yup, do I need to edit anything else?

Thanks :D


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 Post subject:
PostPosted: Sun Mar 01, 2009 12:51 pm 
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Location: Devon, UK
This looks like it might possibly be a bug. Could you please attach as .txt files: the full debug output (with 'debug = 3' in a2billing.conf and 'core set verbose 15' at the Asterisk prompt), a2billing.conf itself, and the relevant sections of extensions.conf.


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 Post subject:
PostPosted: Sun Mar 01, 2009 1:44 pm 
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Joined: Mon Aug 18, 2008 7:02 pm
Posts: 15
here is the information, let me know if anything is missing.

Thanks


Attachments:
debug.txt [32.78 KiB]
Downloaded 501 times
a2billing.txt [14.14 KiB]
Downloaded 501 times
extensions.txt [119 Bytes]
Downloaded 494 times
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 Post subject:
PostPosted: Sun Mar 01, 2009 6:27 pm 
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Location: Devon, UK
Can you please test the following modification:
Edit A2Billing_AGI/a2billing.php around line 947. Change this:
Code:
if ($charge_callback){
to this:
Code:
if ($charge_callback && $RateEngine->answeredtime>0) {


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 Post subject:
PostPosted: Mon Mar 02, 2009 12:21 am 
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Posts: 15
Thanks. Looks good and working fine.

I try to call another mobile of mine which is off, it keep ringing till asterisk shows the following:
[Mar 2 08:18:10] -- Nobody picked up in 60000 ms

Is it possible to lower down the timer?

Thanks


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 Post subject:
PostPosted: Mon Mar 02, 2009 12:43 am 
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Location: Devon, UK
Sure. See 'dialcommand_param' in a2billing.conf, the comments above it and also the documentation for Asterisk's Dial() command.


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 Post subject:
PostPosted: Mon Mar 02, 2009 3:33 am 
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Posts: 15
Thanks :) I set 30s

Added after 1 hours 24 minutes:

one very last qns, if the destination party phone is off, the call keeps ringing, why is this so?

thanks


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 Post subject:
PostPosted: Mon Mar 02, 2009 4:49 pm 
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Location: Devon, UK
Most probably that is due to your carrier not passing the correct signalling back to you. A solid understanding of SIP, 'set sip debug' at the Asterisk console and Wireshark are indispensable when diagnosing issues such as these.


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 Post subject:
PostPosted: Mon Mar 02, 2009 11:21 pm 
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Posts: 15
If I try with softphone, i can hear the operator that tell me that the number is currently unavailable but if I use a2billing, it can't seems to detect :)

any idea? :)

thanks


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 Post subject:
PostPosted: Tue Mar 03, 2009 2:04 am 
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If you'd read the links I gave previously you'd already know that Asterisk doesn't 'detect' the state of the remote party; your carrier must send signalling to inform Asterisk of their disposition. In this instance your carrier should return a "480 Temporarily Unavailable".


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 Post subject:
PostPosted: Tue Mar 03, 2009 11:29 am 
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thanks! i will try to debug myself and let you know.


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 Post subject:
PostPosted: Wed Mar 04, 2009 2:23 am 
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Posts: 42
If I was to hazard a guess, I would say that you are generating a ring tone for the calling party. You need to study the a2billing.conf file probably and understand it. Particularly your "dialcommand_param" variable. You probably have a lower case "r" defined within the variable which will mean that a2billing will generate a ring tone for the calling party.

Try removing the lower case "r" out. It should make a difference. Study the other options as well so you can get a better understanding of this variable.


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 Post subject:
PostPosted: Fri Mar 27, 2009 4:46 pm 
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Joined: Mon Aug 18, 2008 7:02 pm
Posts: 15
Thanks, I will give it a try =)

Sorry as the forum did not send me the email notice.


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