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 Post subject: A2billing passing more/different dialed numbers
PostPosted: Sun Aug 20, 2006 7:36 am 
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Joined: Mon Jul 17, 2006 10:18 pm
Posts: 11
Location: U.S.A.
-- Playing 'digits/10' (language 'en')
a2billing.php: [REQUESTED SetCallerID : 816xxxxxx]
a2billing.php: [EXEC SetCallerID : 816xxxxxx]
a2billing.php: line:518 - UPDATE cc_card SET inuse=inuse+1 WHERE username='9760252662'
a2billing.php: line:166 - [CHANNEL STATUS : 6 = Line is up]
a2billing.php: line:170 - [CREDIT STATUS : 10.00000]
a2billing.php: line:171 - [CREDIT MIN_CREDIT_2CALL : 0]
a2billing.php: line:238 - RES sip_iax_pstndirect_call DTMF : -1
a2billing.php: line:252 - TRUNK - dnid : -1 (1)
a2billing.php: line:543 - 1 && && 2&& 0
-- Playing 'prepaid-enter-dest' (language 'en')
a2billing.php: line:550 - RES DTMF : 01122333480055699969
a2billing.php: line:570 - DESTINATION ::> 01122333480055699969
a2billing.php: line:572 - APPLY_RULES DESTINATION ::> 22333480055699969
a2billing.php: line:609 - ERROR ::> RateEngine didnt succeed to match the dialed number over the ratecard (Please check : id the ratecard is well create ; if the removeInter_Prefix is set according to your prefix in the ratecard ; if you hooked the ratecard to the tariffgroup)
a2billing.php: line:166 - [CHANNEL STATUS : 6 = Line is up]
a2billing.php: line:170 - [CREDIT STATUS : 10.00000]
a2billing.php: line:171 - [CREDIT MIN_CREDIT_2CALL : 0]
a2billing.php: line:238 - RES sip_iax_pstndirect_call DTMF : -1
a2billing.php: line:252 - TRUNK - dnid : -1 (1)
a2billing.php: line:543 - 1 && && 2&& 1
-- Playing 'prepaid-enter-dest' (language 'en')
a2billing.php: line:550 - RES DTMF : 0112348056999969996
a2billing.php: line:570 - DESTINATION ::> 0112348056999969996
a2billing.php: line:572 - APPLY_RULES DESTINATION ::> 2348056999969996
a2billing.php: line:611 - OK - RESFINDRATE::> 1
a2billing.php: line:633 - RES_ALL_CALCULTIMEOUT ::> 1
a2billing.php: line:650 - TIMEOUT::> 2460 : minutes=41 - seconds=0
-- Playing 'digits/40' (language 'en')
-- Playing 'digits/1' (language 'en')
a2billing.php: line:811 - app_callingcard: Dialing 'IAX2/[email protected]/2348056999969996|30|HL(2460000:61000:30000)' with timeout of '2460'.
a2billing.php:
-- AGI Script Executing Application: (Dial) Options: (IAX2/[email protected]/2348056999969996|30|HL(2460000:61000:30000))
-- Limit Data for this call:
-- - timelimit = 2460000
-- - play_warning = 61000
-- - play_to_caller= yes
-- - play_to_callee= no
-- - warning_freq = 30000
-- - start_sound = UNDEF
-- - warning_sound = timeleft
-- - end_sound = UNDEF
-- Called [email protected]/2348056999969996
-- IAX2/voipjet-1 is circuit-busy
-- Hungup 'IAX2/voipjet-1'
== Everyone is busy/congested at this time (1:0/1/0)
a2billing.php: line:885 - FAILOVER app_callingcard: Dialing 'IAX2/[email protected]/2348056999969996|30|HL(2460000:61000:30000)' with timeout of '2460'.
a2billing.php:
-- AGI Script Executing Application: (DIAL) Options: (IAX2/[email protected]/2348056999969996|30|HL(2460000:61000:30000))
-- Limit Data for this call:
-- - timelimit = 2460000
-- - play_warning = 61000
-- - play_to_caller= yes
-- - play_to_callee= no
-- - warning_freq = 30000
-- - start_sound = UNDEF
-- - warning_sound = timeleft
-- - end_sound = UNDEF
-- Called [email protected]/2348056999969996
-- Hungup 'IAX2/voipjet-2'
== Everyone is busy/congested at this time (1:0/0/1)
a2billing.php: line:698 - 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) VALUES ('2156059000.45', 'SIP/38.114.14.210-088c20d0', '9760252662', '', CURRENT_TIMESTAMP - INTERVAL 0 SECOND , '0', '2348056999969996', 'CHANUNAVAIL', now(), '0.243', '-0', '', '', 'NGR', '1', '1', '4', '3', '8163310029', '0')
a2billing.php: line:166 - [CHANNEL STATUS : 6 = Line is up]
a2billing.php: line:170 - [CREDIT STATUS : 10.00000]
a2billing.php: line:171 - [CREDIT MIN_CREDIT_2CALL : 0]
a2billing.php: line:238 - RES sip_iax_pstndirect_call DTMF : -1
a2billing.php: line:252 - TRUNK - dnid : -1 (1)
a2billing.php: line:543 - 1 && && 2&& 2
-- Playing 'prepaid-enter-dest' (language 'en')
a2billing.php: line:550 - RES DTMF : 00112234880235005038
a2billing.php: line:570 - DESTINATION ::> 00112234880235005038
a2billing.php: line:572 - APPLY_RULES DESTINATION ::> 112234880235005038
a2billing.php: line:609 - ERROR ::> RateEngine didnt succeed to match the dialed number over the ratecard (Please check : id the ratecard is well create ; if the removeInter_Prefix is set according to your prefix in the ratecard ; if you hooked the ratecard to the tariffgroup)
a2billing.php: line:166 - [CHANNEL STATUS : 6 = Line is up]
a2billing.php: line:170 - [CREDIT STATUS : 10.00000]
a2billing.php: line:171 - [CREDIT MIN_CREDIT_2CALL : 0]
a2billing.php: line:238 - RES sip_iax_pstndirect_call DTMF : -1
a2billing.php: line:252 - TRUNK - dnid : -1 (1)
a2billing.php: line:543 - 1 && && 2&& 3
-- Playing 'prepaid-enter-dest' (language 'en')
a2billing.php: line:550 - RES DTMF : 01111234488056996999
a2billing.php: line:570 - DESTINATION ::> 01111234488056996999
a2billing.php: line:572 - APPLY_RULES DESTINATION ::> 11234488056996999
a2billing.php: line:609 - ERROR ::> RateEngine didnt succeed to match the dialed number over the ratecard (Please check : id the ratecard is well create ; if the removeInter_Prefix is set according to your prefix in the ratecard ; if you hooked the ratecard to the tariffgroup)
a2billing.php: line:518 - UPDATE cc_card SET inuse=inuse-1 WHERE username='9760252662'
-- AGI Script a2billing.php completed, returning 0


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 Post subject:
PostPosted: Sun Aug 20, 2006 6:55 pm 
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Joined: Mon May 29, 2006 10:38 am
Posts: 50
If I understand your post correctly, it appears that when you dial a number you are getting double and even triple tones/number.

Firs, its not an a2billing issue and it may be an asterisk issue but I think is more of a voip carrier issue.

I have had the same issue before, except it was with sipphone.com/Gizmo. I tried many options for weeks an it just didn't work correctly. Finally I tried broadvoice and that did the trick. Apparently, it has do do with the way dtmf is sent, can't really give a better explanation. If you visit gizmoproject.com you can read the post on this topic.


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 Post subject: PAssing extra numbers
PostPosted: Mon Aug 21, 2006 3:58 pm 
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Joined: Mon Jul 17, 2006 10:18 pm
Posts: 11
Location: U.S.A.
hello rick809,
I checked the dtmf of my asterisk and the provider voipjet and settings seems to be just fine. My concern is with the extra digits that a2billing passes to initiate a call which results to the number dialed is unavailable. May be I'll have to reinstall a2billing and try one more time. If you look at the log you'll see that it passes extra digits.....e.g if you intend to dial 555-5555 it passes 55555-555555 we have a working asterisk that can make and recieve calls without problems any suggestions?


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 Post subject:
PostPosted: Mon Aug 21, 2006 6:31 pm 
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Joined: Mon Jun 19, 2006 5:14 pm
Posts: 218
there seem to be two problems here

1. The DTMF you are getting from whomever is placing the call is bouncing hence you are ending up with double digits. I'll talk to that later

2. You are also placing the call over voipjet without the proper prefix.
Quote:
Called [email protected]/2348056999969996


If memory serves me correct with Voipjet you must use the 011 prefix when you are calling outside the USA/Canada/Caribbean and "1' for usa/canada/caribbean... So your Voipjet trunk definition must add back the 011 if your tariff definition is removing the leading 011 or 00

Back to the DTMF. Your log does not show where you are getting the DTMF or how it is showing up in your server. However it is getting there, it is that trunk's or ATA's context that need to be configured to support way the DTMF is being presented. So I would track down the definition of this context and make sure you have the proper DTMF config:
dtmf=inband,info,rfc2833
to support what the sender is using. FYI IAX2 always uses inband, SIP can use either of the three so u need to configure based on what your sender uses

If you want to test the dial prefix issue separate from the DTMF, use speed dial feature. Define speed dial feature with 1 digit representing the number you want to call ... When prompted for the number you want to call enter the speed dial number followed by # key. It will cause the number you want to dial go get sent with dealing with the DTMF issues and this way you can separate and test solutions to both problems independently


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 Post subject: Thank you much Gue
PostPosted: Mon Aug 21, 2006 8:48 pm 
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Joined: Mon Jul 17, 2006 10:18 pm
Posts: 11
Location: U.S.A.
Thank you much Gue will try it out.
For the voipjet trunk I was using rfc2833 intsead of inband and also will add the prefix 011. I will try it out.
Thanks for your suggestions


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 Post subject:
PostPosted: Mon Aug 21, 2006 9:07 pm 
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Joined: Mon Jun 19, 2006 5:14 pm
Posts: 218
you still did not say how you are getting the digits that you want to dial over the voipjet trunk. It is that connection that is messing up the DTMF. it is that connection that needs addressing. you should also check the physical device to make sure it is not bouncing when you press the keys

for iax trunks u usually don't have to specify the dtmf. if you look at the config spelled out by voipjet you will see they don't even bother telling you how to define the iax dtmf. this is because the iax conf is using the default .


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 Post subject:
PostPosted: Tue Aug 22, 2006 12:04 am 
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Joined: Mon Jul 17, 2006 10:18 pm
Posts: 11
Location: U.S.A.
gue,
Scenario...
I enabled caller ID and did same on the web interface and also selected postpay as opposed to prepaid, so that i dont have to put in numbers, I used my home phone device to call my asterisk box xxx-xxxx I get the prompt you have 10 dollars, enter the number you wish to call....press the numbers 0112658056xxxxxx....while monitoring the log I noticed extra digits are passed on and the log is showing double or tripple digits as opposed to the single number digit dialed. Then I get this number is unavailable and a mismatch error in the log
"RateEngine didnt succeed to match the dialed number over the ratecard (Please check : id the ratecard is well create ; if the removeInter_Prefix is set according to your prefix in the ratecard ; if you hooked the ratecard to the tariffgroup) " .
Am I missing something in the overall install/update?
As regards the device I tried calling via an AtA device and hooked to the same phone and it worked fine without problem. The connection is fine.
If you notice your initial reply to the log I posted,
Called [email protected]/2348056999969996
The 9999 is suppose to be a single digit (9) and not 9999 and sometimes it passes 0011111 as instead of 011. Just wondering


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 Post subject:
PostPosted: Tue Aug 22, 2006 1:33 am 
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Joined: Mon Jun 19, 2006 5:14 pm
Posts: 218
Does your server have a DID that you are calling to get into the a2billing prompt?? Most likely yes because you say you call from your house fone.

Well it is how that trunk is defined that is the source of your DTMF problem.

That incoming trunk/DID is registered with a vendor and they are sending your box the DTMF in a particular fashion inband, info or RFC2833

You need to identify the context where the DID trunk is defined in your sip.conf or iax.conf and set the dtmfmode to match what the DID vendor is sending you.

Quote:
"RateEngine didnt succeed to match the dialed number over the ratecard (Please check : id the ratecard is well create ; if the removeInter_Prefix is set according to your prefix in the ratecard ; if you hooked the ratecard to the tariffgroup) " .
Am I missing something in the overall install/update?


You get this failure message because you are matching on 234 [most likely] but if you look at your dtmf strings you will notice that you have things like 2234 and 112234. So the pattern will not match your 234. When the dtmf is working properly like in the ATA case you will get past the pattern matching state, you will hear the minutes announced but you will fail to dial the number because the 011 is missing from the dial string sent to voipjet.

So you have to fix the dtmf problem and the fix the prefix problem so that 011 is sent to voipjet along with the number you are calling


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VoIP Billing solution


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