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 Post subject: Calls are not going throug
PostPosted: Sun Jun 21, 2009 1:49 am 
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Joined: Thu May 15, 2008 2:02 am
Posts: 115
Hello everybody
Having problems with the calls but i think is something wrong I did
could you please help me out to find out what is wrong please
thank you iam using the v 1.3
regards

a2billing.php:
a2billing.php: file:Class.A2Billing.php - line:621 - get_agi_request_parameter = 955983 ; SIP/091943-0edf2430 ; 1245514358.1688 ; 091943 ; 17864971516
a2billing.php: file:a2billing.php - line:145 - [NO ANSWER CALL]
a2billing.php: file:Class.A2Billing.php - line:1640 - SELECT credit, tariff, activated, inuse, simultaccess, typepaid, creditlimit, language, removeinterprefix, redial, enableexpire, UNIX_TIMESTAMP(expirationdate), expiredays, nbused, UNIX_TIMESTAMP(firstusedate), UNIX_TIMESTAMP(cc_card.creationdate), cc_card.currency, cc_card.lastname, cc_card.firstname, cc_card.email, cc_card.uipass, cc_card.id_campaign, cc_card.id, useralias FROM cc_card LEFT JOIN cc_tariffgroup ON tariff=cc_tariffgroup.id WHERE username='091943'
[Jun 20 09:12:38] ERROR[5662]: pbx.c:2793 ast_func_write: Function LANGUAGE not registered
a2billing.php: file:Class.A2Billing.php - line:1714 - [SET LANGUAGE() en]
a2billing.php: file:Class.A2Billing.php - line:654 - [CARD STATUS UPDATE : UPDATE cc_card SET inuse=inuse+1 WHERE username='091943']
a2billing.php: file:Class.A2Billing.php - line:1400 - [AUTO SetCallerID]
a2billing.php: file:Class.A2Billing.php - line:1406 - [REQUESTED SetCallerID : 955983]
a2billing.php: file:Class.A2Billing.php - line:1417 - [EXEC SetCallerID : 955983]
a2billing.php: file:a2billing.php - line:172 - [CHANNEL STATUS : 4 = Line is ringing]
a2billing.php: file:a2billing.php - line:173 - [CREDIT : 10.00000][CREDIT MIN_CREDIT_2CALL : 0]
a2billing.php: file:Class.A2Billing.php - line:676 - 1 && && 11&& 0
a2billing.php: file:Class.A2Billing.php - line:701 - DESTINATION ::> 17864971516
a2billing.php: file:Class.A2Billing.php - line:703 - RULES APPLY ON DESTINATION ::> 17864971516
a2billing.php: file:Class.A2Billing.php - line:741 - OK - RESFINDRATE::> 1
a2billing.php: file:Class.A2Billing.php - line:763 - RES_ALL_CALCULTIMEOUT ::> 1
a2billing.php: file:Class.A2Billing.php - line:780 - TIMEOUT::> 76923 : minutes=1282 - seconds=3
a2billing.php: file:Class.RateEngine.php - line:1012 - app_callingcard: Dialing 'SIP/sbc01-car.dal.us.icall.net/17864971516|60|HRgrL(76923000:61000:30000)' with timeout of '76923'.
a2billing.php:
a2billing.php: file:Class.RateEngine.php - line:1037 - app_callingcard: CIDGROUPID='-1' OUTBOUND CID SELECTED IS '0'.
-- AGI Script Executing Application: (Dial) Options: (SIP/sbc01-car.dal.us.icall.net/17864971516|60|HRgrL(76923000:61000:30000))
== Using SIP RTP CoS mark 5
-- Called sbc01-car.dal.us.icall.net/17864971516|60|HRgrL(76923000:61000:30000)
-- SIP/sbc01-car.dal.us.icall.net-ac0a99d0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
a2billing.php: file:Class.RateEngine.php - line:1148 - [USEDRATECARD - FAIL =0]
a2billing.php: 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 ('1245514358.1688', 'SIP/091943-0edf2430', '091943', '', CURRENT_TIMESTAMP - INTERVAL 0 SECOND , '0', '17864971516', 'CONGESTION', now(), '0.0078', '-0', '', '', 'usa', '1', '1', '1', '2', '955983', '0', '0.0055', '0', '0')]
a2billing.php: file:Class.RateEngine.php - line:902 - [CC_asterisk_stop 1.1: SQL: DONE : result=1]
a2billing.php: file:a2billing.php - line:312 - [a2billing account stop]
a2billing.php: file:Class.A2Billing.php - line:654 - [CARD STATUS UPDATE : UPDATE cc_card SET inuse=inuse-1 WHERE username='091943']
-- <SIP/091943-0edf2430>AGI Script a2billing.php completed, returning -1
== Using SIP RTP CoS mark 5


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 Post subject: Re: Calls are not going throug
PostPosted: Sun Jun 21, 2009 12:05 pm 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
Check first that you can call sbc01-car.dal.us.icall.net/17864971516 without the complexities of A2Billng, direct from Asterisk to prove out your trunk.

Joe


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 Post subject: Re: Calls are not going throug
PostPosted: Sun Jun 21, 2009 5:50 pm 
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Joined: Thu May 15, 2008 2:02 am
Posts: 115
Hello
Thank you for for the answer
but i forgot iam calling to that provider actully and everthing is fine from asterisk
Thank you


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 Post subject: Re: Calls are not going throug
PostPosted: Wed Jun 24, 2009 3:23 pm 
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Joined: Thu May 15, 2008 2:02 am
Posts: 115
Hello can anybody help me out please

I can make calls with asterisk without any problem everything is fine, but when I try to call throug A2B i get the above messages any idea please

thank you so much


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 Post subject: Re: Calls are not going throug
PostPosted: Wed Jun 24, 2009 6:36 pm 
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Joined: Thu May 15, 2008 2:02 am
Posts: 115
Hello I was doing some changes in my config
and now i got this error but i have that trunk in my sip.conf file and i dont why is this message
thank you so much for your help


a2billing.php: file:Class.RateEngine.php - line:1037 - app_callingcard: CIDGROUPID='-1' OUTBOUND CID SELECTED IS '0'.
-- AGI Script Executing Application: (Dial) Options: (SIP/17864971516@icall|60|HRgrL(76923000:61000:30000))
== Using SIP RTP CoS mark 5
[Jun 24 04:33:25] WARNING[10105]: chan_sip.c:4224 create_addr: No such host: icall|60|HRgrL(76923000
[Jun 24 04:33:25] WARNING[10105]: app_dial.c:1468 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
a2billing.php: file:Class.RateEngine.php - line:1148 - [USEDRATECARD - FAIL =0]
a2billing.php: 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 ('1245843204.78650', 'SIP/091943-c864ad40', '091943', '', CURRENT_TIMESTAMP - INTERVAL 0 SECOND , '0', '17864971516', 'CHANUNAVAIL', now(), '0.0078', '-0', '', '', 'usa', '1', '1', '1', '2', '955983', '0', '0.0055', '0', '0')]
a2billing.php: file:Class.RateEngine.php - line:902 - [CC_asterisk_stop 1.1: SQL: DONE : result=1]
a2billing.php: file:a2billing.php - line:312 - [a2billing account stop]
a2billing.php: file:Class.A2Billing.php - line:654 - [CARD STATUS UPDATE : UPDATE cc_card SET inuse=inuse-1 WHERE username='091943']
-- <SIP/091943-c864ad40>AGI Script a2billing.php completed, returning -1


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 Post subject: Re: Calls are not going throug
PostPosted: Wed Jun 24, 2009 6:55 pm 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
COuld this be a problem?

chan_sip.c:4224 create_addr: No such host: icall


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 Post subject: Re: Calls are not going throug
PostPosted: Wed Jun 24, 2009 7:10 pm 
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Joined: Thu May 15, 2008 2:02 am
Posts: 115
ok thank you for the answer

but iam calling without problems to that provider from asterisk
the problem is only when iam trying to go throug A2B

this my sip.com

[icall]
type=friend
host=sbc01-car.dal.us.icall.net
context=icall_in
username=xxxx
secret=xxxxx
dtmfmode=rfc2833
nat=yes
disallow=all
allow=g729
allow=ulaw
allow=alaw
allow=gsm

This is my trunk in my sip
If I make a call only with asterisk to that provider works fine

Any idea??
Thanks


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 Post subject: Re: Calls are not going throug
PostPosted: Wed Jun 24, 2009 7:20 pm 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
WHat do the fields say in your A2Billing trunk?


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 Post subject: Re: Calls are not going throug
PostPosted: Wed Jun 24, 2009 7:28 pm 
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Joined: Thu May 15, 2008 2:02 am
Posts: 115
This my trunk
VOIP-PROVIDER : ICALL
LABEL icall-outbound
PROVIDER TECH : SIP
PROVIDER IP : icall


Thanks


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 Post subject: Re: Calls are not going throug
PostPosted: Sat Jun 27, 2009 3:19 pm 
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Joined: Thu May 15, 2008 2:02 am
Posts: 115
Hello Everybody

Can somebody help me please
I can not make calls this is the message that I got from /var/log/asterisk/

[Jun 27 01:07:30] ERROR[17331] pbx.c: Function LANGUAGE not registered
[Jun 27 01:07:31] WARNING[17331] chan_sip.c: No such host: voipms|60|HRgrL(76923000
[Jun 27 01:07:31] WARNING[17331] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)

this is my sip.conf

[voipms]
canreinvite=no
context=mycontext
host=sip.us4.voip.ms
secret= xxxxx
type=friend
username=xxxxx
disallow=all
allow=ulaw
allow=g729 ; Uncomment if you support G729
fromuser=xxxxxx
trustrpid=yes
dtmfmode=info
qualify=yes
sendrpid=yes
insecure=port,invite
nat=yes ; Uncomment this if your box is behind a NAT

This is my trunk in A2B

VOIP-PROVIDER = voip

LABEL =international

ADD PREFIX =

REMOVE PREFIX =

PROVIDER TECH =SIP

PROVIDER IP = voipms

ADDITIONAL PARAMETER =

FAILOVER TRUNK

I can make calls with asterisk to that provider without problems

This is the cli:

a2billing.php: file:Class.A2Billing.php - line:621 - get_agi_request_parameter = 513227 ; SIP/604600-07bd9ac0 ; 1246089935.131441 ; 604600 ; 18883396699
== Spawn extension (icall-outbound, 19375489875, 1) exited non-zero on 'SIP/5060-07c0a890'
a2billing.php: file:a2billing.php - line:145 - [NO ANSWER CALL]
a2billing.php: file:Class.A2Billing.php - line:1640 - SELECT credit, tariff, activated, inuse, simultaccess, typepaid, creditlimit, language, removeinterprefix, redial, enableexpire, UNIX_TIMESTAMP(expirationdate), expiredays, nbused, UNIX_TIMESTAMP(firstusedate), UNIX_TIMESTAMP(cc_card.creationdate), cc_card.currency, cc_card.lastname, cc_card.firstname, cc_card.email, cc_card.uipass, cc_card.id_campaign, cc_card.id, useralias FROM cc_card LEFT JOIN cc_tariffgroup ON tariff=cc_tariffgroup.id WHERE username='604600'
[Jun 27 01:05:36] ERROR[17204]: pbx.c:2793 ast_func_write: Function LANGUAGE not registered
a2billing.php: file:Class.A2Billing.php - line:1714 - [SET LANGUAGE() en]
a2billing.php: file:Class.A2Billing.php - line:654 - [CARD STATUS UPDATE : UPDATE cc_card SET inuse=inuse+1 WHERE username='604600']
a2billing.php: file:Class.A2Billing.php - line:1400 - [AUTO SetCallerID]
a2billing.php: file:Class.A2Billing.php - line:1406 - [REQUESTED SetCallerID : 513227]
a2billing.php: file:Class.A2Billing.php - line:1417 - [EXEC SetCallerID : 513227]
a2billing.php: file:a2billing.php - line:172 - [CHANNEL STATUS : 4 = Line is ringing]
a2billing.php: file:a2billing.php - line:173 - [CREDIT : 10.00000][CREDIT MIN_CREDIT_2CALL : 0]
a2billing.php: file:Class.A2Billing.php - line:676 - 1 && && 11&& 0
a2billing.php: file:Class.A2Billing.php - line:701 - DESTINATION ::> 18883396699
a2billing.php: file:Class.A2Billing.php - line:703 - RULES APPLY ON DESTINATION ::> 18883396699
a2billing.php: file:Class.A2Billing.php - line:741 - OK - RESFINDRATE::> 1
a2billing.php: file:Class.A2Billing.php - line:763 - RES_ALL_CALCULTIMEOUT ::> 1
a2billing.php: file:Class.A2Billing.php - line:780 - TIMEOUT::> 76923 : minutes=1282 - seconds=3
a2billing.php: file:Class.RateEngine.php - line:1012 - app_callingcard: Dialing 'SIP/18883396699@voipms|60|HRgrL(76923000:61000:30000)' with timeout of '76923'.
a2billing.php:
a2billing.php: file:Class.RateEngine.php - line:1037 - app_callingcard: CIDGROUPID='-1' OUTBOUND CID SELECTED IS '0'.
-- AGI Script Executing Application: (Dial) Options: (SIP/18883396699@voipms|60|HRgrL(76923000:61000:30000))
== Using SIP RTP CoS mark 5
[Jun 27 01:05:36] WARNING[17204]: chan_sip.c:4224 create_addr: No such host: voipms|60|HRgrL(76923000
[Jun 27 01:05:36] WARNING[17204]: app_dial.c:1468 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
a2billing.php: file:Class.RateEngine.php - line:1148 - [USEDRATECARD - FAIL =0]
a2billing.php: 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 ('1246089935.131441', 'SIP/604600-07bd9ac0', '604600', '', CURRENT_TIMESTAMP - INTERVAL 0 SECOND , '0', '18883396699', 'CHANUNAVAIL', now(), '0.0078', '-0', '', '', 'usa', '5', '5', '5', '6', '513227', '0', '0.0055', '0', '0')]
a2billing.php: file:Class.RateEngine.php - line:902 - [CC_asterisk_stop 1.1: SQL: DONE : result=1]
a2billing.php: file:a2billing.php - line:312 - [a2billing account stop]



Can somebody help me please

Thank you
Regards
a2billing.php: file:Class.A2Billing.php - line:654 - [CARD STATUS UPDATE : UPDATE cc_card SET inuse=inuse-1 WHERE username='604600']
-- <SIP/604600-07bd9ac0>AGI Script a2billing.php completed, returning -1


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 Post subject: Re: Calls are not going throug
PostPosted: Sat Jun 27, 2009 5:19 pm 
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Joined: Thu May 15, 2008 2:02 am
Posts: 115
Never mind
I just solved it
Thank you for your help

Regards


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 Post subject: Re: Calls are not going throug
PostPosted: Wed Mar 17, 2010 8:27 pm 
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Joined: Wed Mar 17, 2010 8:20 pm
Posts: 4
would you tell me how you fix it, because i am gatting the same probleme.
i also use VoIPms as trunk


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 Post subject: Re: Calls are not going throug
PostPosted: Thu Mar 18, 2010 2:39 pm 
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Joined: Thu May 15, 2008 2:02 am
Posts: 115
if you are using asterisk 1.6 you need to change "," instead of "|" on Dialcommand param in agi.conf.

thanks


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 Post subject: Re: Calls are not going throug
PostPosted: Thu Mar 18, 2010 5:24 pm 
Offline

Joined: Wed Mar 17, 2010 8:20 pm
Posts: 4
thank you for the replay.
i have lookef at all my config files, i didn r see agi.conf
maybe it is in an specific place, can you tell me where it is, please.

I pasted what asterisk CLI show when i place a call


-- Executing ["my DID"@from-trunk:1] Set("SIP/voipms1-00000065", "__FROM_DID="my DID"") in new stack
-- Executing ["my DID"@from-trunk:2] Gosub("SIP/voipms1-00000065", "app-blacklist-check,s,1") in new stack
-- Executing [s@app-blacklist-check:1] GotoIf("SIP/voipms1-00000065", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:2] Return("SIP/voipms1-00000065", "") in new stack
-- Executing ["my DID"@from-trunk:3] ExecIf("SIP/voipms1-00000065", "0 ?Set(CALLERID(name)=15143444562)") in new stack
-- Executing ["my DID"@from-trunk:4] SetMusicOnHold("SIP/voipms1-00000065", "acc_1") in new stack
-- Executing ["my DID"@from-trunk:5] Set("SIP/voipms1-00000065", "__MOHCLASS=acc_1") in new stack
-- Executing ["my DID"@from-trunk:6] Set("SIP/voipms1-00000065", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing ["my DID"@from-trunk:7] Set("SIP/voipms1-00000065", "CALLERPRES()=allowed_not_screened") in new stack
-- Executing ["my DID"@from-trunk:8] Goto("SIP/voipms1-00000065", "custom-a2billing,"my DID",1") in new stack
-- Goto (custom-a2billing,"my DID",1)
-- Executing ["my DID"@custom-a2billing:1] Answer("SIP/voipms1-00000065", "") in new stack
-- Executing ["my DID"@custom-a2billing:2] Wait("SIP/voipms1-00000065", "1") in new stack
-- Executing ["my DID"@custom-a2billing:3] DeadAGI("SIP/voipms1-00000065", "a2billing.php,1") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
-- <SIP/voipms1-00000065> Playing 'prepaid-enter-pin-number.gsm' (language 'en')
-- Playing 'prepaid-you-have' (escape_digits=#) (sample_offset 0)
-- <SIP/voipms1-00000065> Playing 'digits/2.gsm' (language 'en')
-- Playing 'dollars' (escape_digits=#) (sample_offset 0)
-- <SIP/voipms1-00000065> Playing 'prepaid-enter-dest.gsm' (language 'en')
-- Playing 'prepaid-you-have' (escape_digits=#) (sample_offset 0)
-- <SIP/voipms1-00000065> Playing 'digits/90.gsm' (language 'en')
-- Playing 'prepaid-minutes' (escape_digits=#) (sample_offset 0)
-- AGI Script Executing Application: (DIAL) Options: (SIP/[email protected]|60|HRrL(5400000:61000:30000))
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
== Everyone is busy/congested at this time (1:0/0/1)
-- Playing 'prepaid-dest-unreachable' (escape_digits=#) (sample_offset 0)
-- <SIP/voipms1-00000065> Playing 'prepaid-enter-dest.gsm' (language 'en')
== Spawn extension (custom-a2billing, "my DID", 3) exited non-zero on 'SIP/voipms1-00000065'


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 Post subject: Re: Calls are not going throug
PostPosted: Thu Mar 18, 2010 6:00 pm 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
Quote:
if you are using asterisk 1.6 you need to change "," instead of "|" on Dialcommand param in agi.conf.


Not true, just change the global parameter asterisk version from 1_4 to 1_6

Joe


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