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 Post subject: outbound error
PostPosted: Tue May 20, 2008 1:19 am 
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Joined: Tue May 20, 2008 1:12 am
Posts: 22
Hi I am a new bee here!

sucessfully installed Teixbox and a2billing. I have created one truck with a SIP provider and routed the DID to a2billing. when I dial the numner it asked for the pin and say the remaining $$s and the minutes. but the call did not go thrugh.. it ring once and say the customer is unavaiilable.

Please advice! i attach the a2billing log here. I appreciate anybody's help here.
Thanks.

LEFT JOIN cc_package_offer ON cc_package_offer.id=cc_tariffgroup.id_cc_package_offer
WHERE (dialprefix=SUBSTRING('14165695266',1,length(dialprefix)) OR dialprefix='defaultprefix')
AND startingdate<= CURRENT_TIMESTAMP AND (expirationdate > CURRENT_TIMESTAMP OR expirationdate IS NULL OR LENGTH(expirationdate)<5)
AND startdate<= CURRENT_TIMESTAMP AND (stopdate > CURRENT_TIMESTAMP OR stopdate IS NULL OR LENGTH(stopdate)<5)
AND (starttime <= 1244 AND endtime >=1244)
AND idtariffgroup='3'
AND ( dnidprefix=SUBSTRING('felixjoe',1,length(dnidprefix)) OR (dnidprefix='all' AND 0 = 0))
AND ( calleridprefix=SUBSTRING('9057921343',1,length(calleridprefix)) OR (calleridprefix='all' AND 0 = 0))
ORDER BY LENGTH(dialprefix) DESC
]
[19/05/2008 20:44:34]:[file:Class.RateEngine.php - line:192]:[CallerID:9057921343]:[CN:5388980051]:[[rate-engine: Count Total result 1]]
[19/05/2008 20:44:34]:[file:Class.RateEngine.php - line:281]:[CallerID:9057921343]:[CN:5388980051]:[[CC_asterisk_rate-engine: Count Total result 1]]
[19/05/2008 20:44:34]:[file:Class.RateEngine.php - line:282]:[CallerID:9057921343]:[CN:5388980051]:[[CC_asterisk_rate-engine: number_trunk 1]]
[19/05/2008 20:44:34]:[file:Class.A2Billing.php - line:736]:[CallerID:9057921343]:[CN:5388980051]:[OK - RESFINDRATE::> 1]
[19/05/2008 20:44:34]:[file:Class.RateEngine.php - line:295]:[CallerID:9057921343]:[CN:5388980051]:[[CC_RATE_ENGINE_ALL_CALCULTIMEOUT (4.15000)]]
[19/05/2008 20:44:34]:[file:Class.RateEngine.php - line:301]:[CallerID:9057921343]:[CN:5388980051]:[[CC_RATE_ENGINE_ALL_CALCULTIMEOUT: k=0 - res_calcultimeout:11850]]
[19/05/2008 20:44:34]:[file:Class.A2Billing.php - line:758]:[CallerID:9057921343]:[CN:5388980051]:[RES_ALL_CALCULTIMEOUT ::> 1]
[19/05/2008 20:44:34]:[file:Class.A2Billing.php - line:775]:[CallerID:9057921343]:[CN:5388980051]:[TIMEOUT::> 11850 : minutes=197 - seconds=30]
[19/05/2008 20:44:41]:[file:Class.RateEngine.php - line:1026]:[CallerID:9057921343]:[CN:5388980051]:[app_callingcard: Dialing 'SIP/vbuzzer.com/14165695266|60|HRgrL(11850000:61000:30000)' with timeout of '11850'.
]
[19/05/2008 20:44:41]:[file:Class.RateEngine.php - line:1051]:[CallerID:9057921343]:[CN:5388980051]:[app_callingcard: CIDGROUPID='-1' OUTBOUND CID SELECTED IS '0'.]
[19/05/2008 20:44:41]:[file:Class.RateEngine.php - line:1056]:[CallerID:9057921343]:[CN:5388980051]:[DIAL SIP/vbuzzer.com/14165695266|60|HRgrL(11850000:61000:30000)]
[19/05/2008 20:44:41]:[file:Class.RateEngine.php - line:1079]:[CallerID:9057921343]:[CN:5388980051]:[[K=0]:[ANSWEREDTIME=0-DIALSTATUS=CONGESTION]]
[19/05/2008 20:44:41]:[file:Class.RateEngine.php - line:1119]:[CallerID:9057921343]:[CN:5388980051]:[FAILOVER app_callingcard: Dialing 'SIP/vbuzzer.com/14165695266|60|HRgrL(11850000:61000:30000)' with timeout of '11850'.
]
[19/05/2008 20:44:41]:[file:Class.RateEngine.php - line:1122]:[CallerID:9057921343]:[CN:5388980051]:[DIAL FAILOVER SIP/vbuzzer.com/14165695266|60|HRgrL(11850000:61000:30000)]
[19/05/2008 20:44:41]:[file:Class.RateEngine.php - line:1129]:[CallerID:9057921343]:[CN:5388980051]:[[FAILOVER K=0]:[ANSTIME=-DIALSTATUS=CONGESTION]]
[19/05/2008 20:44:41]:[file:Class.RateEngine.php - line:1079]:[CallerID:9057921343]:[CN:5388980051]:[[K=0]:[ANSWEREDTIME=0-DIALSTATUS=CONGESTION]]
[19/05/2008 20:44:41]:[file:Class.RateEngine.php - line:1119]:[CallerID:9057921343]:[CN:5388980051]:[FAILOVER app_callingcard: Dialing 'SIP/vbuzzer.com/14165695266|60|HRgrL(11850000:61000:30000)' with timeout of '11850'.
]
[19/05/2008 20:44:41]:[file:Class.RateEngine.php - line:1122]:[CallerID:9057921343]:[CN:5388980051]:[DIAL FAILOVER SIP/vbuzzer.com/14165695266|60|HRgrL(11850000:61000:30000)]
[19/05/2008 20:44:41]:[file:Class.RateEngine.php - line:1129]:[CallerID:9057921343]:[CN:5388980051]:[[FAILOVER K=0]:[ANSTIME=-DIALSTATUS=CONGESTION]]
[19/05/2008 20:44:41]:[file:Class.RateEngine.php - line:1079]:[CallerID:9057921343]:[CN:5388980051]:[[K=0]:[ANSWEREDTIME=0-DIALSTATUS=CONGESTION]]
[19/05/2008 20:44:41]:[file:Class.RateEngine.php - line:1119]:[CallerID:9057921343]:[CN:5388980051]:[FAILOVER app_callingcard: Dialing 'SIP/vbuzzer.com/14165695266|60|HRgrL(11850000:61000:30000)' with timeout of '11850'.
]
[19/05/2008 20:44:41]:[file:Class.RateEngine.php - line:1122]:[CallerID:9057921343]:[CN:5388980051]:[DIAL FAILOVER SIP/vbuzzer.com/14165695266|60|HRgrL(11850000:61000:30000)]
[19/05/2008 20:44:41]:[file:Class.RateEngine.php - line:1129]:[CallerID:9057921343]:[CN:5388980051]:[[FAILOVER K=0]:[ANSTIME=-DIALSTATUS=CONGESTION]]
[19/05/2008 20:44:41]:[file:Class.RateEngine.php - line:1162]:[CallerID:9057921343]:[CN:5388980051]:[[USEDRATECARD - FAIL =0]]
[19/05/2008 20:44:45]:[file:Class.RateEngine.php - line:890]:[CallerID:9057921343]:[CN:5388980051]:[[CC_RATE_ENGINE_UPDATESYSTEM: usedratecard K=0 - (sessiontime=0 :: dialstatus=CONGESTION :: buycost=0 :: cost= : signe_cc_call=-: signe=+)]]
[19/05/2008 20:44:45]:[file:Class.RateEngine.php - line:914]:[CallerID:9057921343]:[CN:5388980051]:[[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 ('2211244230.46', 'SIP/felixjoe-08336518', '5388980051', '', CURRENT_TIMESTAMP - INTERVAL 0 SECOND , '0', '14165695266', 'CONGESTION', now(), '0.02', '-0', '', '', 'ca', '3', '3', '3', '5', '9057921343', '0', '0.01', '0', '0')]]
[19/05/2008 20:44:45]:[file:Class.RateEngine.php - line:917]:[CallerID:9057921343]:[CN:5388980051]:[[CC_asterisk_stop 1.1: SQL: DONE : result=1]]
[19/05/2008 20:44:45]:[file:a2billing.php - line:310]:[CallerID:9057921343]:[CN:5388980051]:[[a2billing account stop]]
[19/05/2008 20:44:45]:[file:a2billing.php - line:170]:[CallerID:9057921343]:[CN:5388980051]:[[CHANNEL STATUS : 6 = Line is up]]
[19/05/2008 20:44:45]:[file:a2billing.php - line:171]:[CallerID:9057921343]:[CN:5388980051]:[[CREDIT : 4.15000][CREDIT MIN_CREDIT_2CALL : 0]]
[19/05/2008 20:44:45]:[file:Class.A2Billing.php - line:671]:[CallerID:9057921343]:[CN:5388980051]:[0 && && 8&& 2]
[19/05/2008 20:44:49]:[file:Class.A2Billing.php - line:678]:[CallerID:9057921343]:[CN:5388980051]:[RES DTMF : -1]
[19/05/2008 20:44:49]:[file:Class.A2Billing.php - line:696]:[CallerID:9057921343]:[CN:5388980051]:[DESTINATION ::> -1]
[19/05/2008 20:44:49]:[file:Class.A2Billing.php - line:698]:[CallerID:9057921343]:[CN:5388980051]:[RULES APPLY ON DESTINATION ::> -1]
[19/05/2008 20:44:49]:[file:Class.A2Billing.php - line:649]:[CallerID:9057921343]:[CN:5388980051]:[[CARD STATUS UPDATE : UPDATE cc_card SET inuse=inuse-1 WHERE username='5388980051']]
[19/05/2008 20:44:49]:[CallerID:9057921343]:[CN:5388980051]:[[exit]]


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 Post subject:
PostPosted: Tue May 20, 2008 1:27 am 
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Joined: Thu Jun 22, 2006 2:19 pm
Posts: 2890
Location: Devon, UK
As you can see A2Billing dialled 3 times, receiving congestion from vbuzzer each time. Perhaps they want you to prefix your calls with 00 or 011?


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 Post subject:
PostPosted: Tue May 20, 2008 1:54 am 
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Joined: Tue May 20, 2008 1:12 am
Posts: 22
Hi Thanks for your prompt reply. I appreciate that.

I have prefixed with '1' here. only a2billing could not get through. Howver, if I dial a number with prefix '1' from my extensions it go through. if that is the problem even the calls from extensions will also fail? right? i dod not know,

please advice.

Thanks
joe


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 Post subject:
PostPosted: Tue May 20, 2008 2:51 am 
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Posts: 2890
Location: Devon, UK
Not necessarily. It may be something other than the dialstring which is causing the call to fail; perhaps you are using different codecs when you place A2Billing in the call path too?
Whatever the problem is, if you consult the Asterisk console with a high verbosity level whilst the call fails you might get an additional clue.


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 Post subject:
PostPosted: Tue May 20, 2008 3:32 am 
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Joined: Tue May 20, 2008 1:12 am
Posts: 22
hi thanks again!

I have captured the asterisk log and attacehd below. Does this give you any additional information to help me out? please note that the call get through, and the $$ amt messagae and the minutes information follow. I could hear a ring and the call aborted saying, the number is unavailable.

Thanks
Joe

v=0
o=root 8766 8766 IN IP4 192.168.1.66
s=session
c=IN IP4 192.168.1.66
t=0 0
m=audio 10770 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
-- Called vbuzzer.com/14165695266
trixbox1*CLI>
<--- SIP read from 209.47.41.24:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bK7f53061f;rport
From: "9057921343" <sip:[email protected]>;tag=as61a71de3
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length:0


<------------->
--- (7 headers 0 lines) ---
trixbox1*CLI>
<--- SIP read from 209.47.41.24:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bK7f53061f;rport
From: "9057921343" <sip:[email protected]>;tag=as61a71de3
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length:0
WWW-Authenticate:Digest realm="vbuzzer.com", nonce="4d5449784d5449314e6a6b334f413d3d", algorithm=MD5
Warning:392 209.47.41.24:80 "Noisy feedback tells: pid=93190 req_src_ip=192.168.1.66 req_src_port=64205 in_uri=sip:sip:[email protected] out_uri=sip:sip:1416569[email protected] via_cnt==1"


<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 209.47.41.24:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.66:5060;branch=z9hG4bK7f53061f;rport
From: "9057921343" <sip:[email protected]>;tag=as61a71de3
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
-- SIP/vbuzzer.com-08362478 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
Really destroying SIP dialog '[email protected]' Method: INVITE
a2billing.php: file:Class.RateEngine.php - line:1129 - [FAILOVER K=0]:[ANSTIME=-DIALSTATUS=CONGESTION]
a2billing.php: file:Class.RateEngine.php - line:1079 - [K=0]:[ANSWEREDTIME=0-DIALSTATUS=CONGESTION]
a2billing.php: file:Class.RateEngine.php - line:1119 - FAILOVER app_callingcard: Dialing 'SIP/vbuzzer.com/14165695266|60|HRgrL(11850000:61000:30000)' with timeout of '11850'.
<snip>


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 Post subject:
PostPosted: Tue May 20, 2008 3:47 am 
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Posts: 2890
Location: Devon, UK
If you're going to show us large debug logs please attach them as a .txt file in future; it saves lots of bandwidth and scrolling.

If you read the remaining parts of your last post you should see the problem: vbuzzer.com are rejecting your call as you've failed to authenticate to them.

The most likely cause is missing credentials in their peer definition in sip.conf. It's also possible they're failing the call because you're offering only μ-law and a-law whilst they perhaps insist on G.729, or some other codec.


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 Post subject:
PostPosted: Tue May 20, 2008 4:49 am 
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Joined: Tue May 20, 2008 1:12 am
Posts: 22
Hi thanks for you reply.

I am sorry that i atatched the file in the post; hereafter I'll attached it. i do not no the rules. sorry again.

Back to the problem; l checked my Sip file and I have added allow all
as follows. the problem still persisits. plz advice thanks.

joe

allow=ulaw&g729&all&all
canreinvite=no
context=from-pstn
dtmf=rfc2833
dtmfmode=rfc2833
host=vbuzzer.com
port=80
username=xxxxxx
secret=xxxxxxx
insecure=very
nat=route
restrictcid=no
type=peer


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 Post subject:
PostPosted: Tue May 20, 2008 5:17 am 
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Location: Devon, UK
The clue here is that the peer definition declares port 80, but the SIP trace shows the standard port 5060 being used; it doesn't look like Asterisk is consulting the peer definition at all, which is why it doesn't respond to the challenge to authenticate with its credentials. I'm guessing here, but I can imagine how this problem might be caused by your use of the deprecated 'insecure=very'. Try 'insecure=port' (or perhaps 'port, invite') instead.

If that doesn't help, my next guess would be to try renaming your peer definition so it doesn't look quite so much like a valid domain name. Don't forget to amend any associated A2Billing trunk definitions to use the new name in the provider IP field.

This is starting to get very off-topic for this forum; we don't support Asterisk itself here, only A2Billing.


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 Post subject:
PostPosted: Wed May 21, 2008 12:55 am 
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Joined: Tue May 20, 2008 1:12 am
Posts: 22
Hi
Thanks for your mail,
I tried all the suggestions you made to me. I still could not solve it. I have attached the log. its still the same as "SIP/2.0 401 Unauthorized"

Please help me to sort this out. I am behind the firewall and enabled the 5060 & 10000 - 20000 on my speedtouch router.

Thanks,
Joe


Attachments:
astrisk log.txt [8.87 KiB]
Downloaded 338 times
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 Post subject:
PostPosted: Wed May 21, 2008 6:30 am 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
Hi

I would suggest ignoring A2Billing for the moment, and concentrate on getting your trunk to work successfully via trixbox.

Set up a freepbx extension, and set up the trunk in FreePBX, make an outbound route, then test. Once you have it working, then whatever name you have for your trunk in FreePBX, copy and paste it into A2Billing trunk in the IPaddress field, then test in A2Billing.

Joe


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 Post subject:
PostPosted: Wed May 21, 2008 7:13 pm 
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Joined: Tue May 20, 2008 1:12 am
Posts: 22
Hi Thanks for your suggestion.

please note that the trunk is working successfully via trixbox.

I can dial from a freepbx extension, and set up the trunk in FreePBX, using theoutbound route.

Thanks


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 Post subject:
PostPosted: Wed May 21, 2008 7:40 pm 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
Then it will work successfully in A2Billing, simply copy and paste the trunk name from FreePBX to the IP address field in A2Billing, set technology type as SIP, and provided you have a rate for the destination, it will work.

Joe


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 Post subject:
PostPosted: Thu May 22, 2008 1:10 am 
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Joined: Tue May 20, 2008 1:12 am
Posts: 22
hi, thanks!

i could fix the SIP 401 error, Now this is my asterix log look like. Please not it say the $$ amt and after dialing the minutes, one ring and go as "the number id ont available" Please help

the problem is DIALSTATUS=CONGESTION

thanks
joe


a2billing.php|1: file:Class.A2Billing.php - line:775 - TIMEOUT::> 11850 : minutes=197 - seconds=30
-- Playing 'prepaid-you-have' (escape_digits=#) (sample_offset 0)
-- <SIP/felixjoe-0a09dc58> Playing 'digits/1' (language 'en')
-- <SIP/felixjoe-0a09dc58> Playing 'digits/hundred' (language 'en')
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
-- <SIP/felixjoe-0a09dc58> Playing 'digits/90' (language 'en')
== Manager 'admin' logged off from 127.0.0.1
-- <SIP/felixjoe-0a09dc58> Playing 'digits/7' (language 'en')
-- Playing 'prepaid-minutes' (escape_digits=#) (sample_offset 0)
-- Playing 'vm-and' (escape_digits=#) (sample_offset 0)
-- <SIP/felixjoe-0a09dc58> Playing 'digits/30' (language 'en')
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
-- Playing 'prepaid-seconds' (escape_digits=#) (sample_offset 0)
a2billing.php|1: file:Class.RateEngine.php - line:1026 - app_callingcard: Dialing 'SIP/vbuzzer.com/14168805266|60|HRgrL(11850000:61000:30000)' with timeout of '11850'.
a2billing.php|1:
a2billing.php|1: file:Class.RateEngine.php - line:1051 - app_callingcard: CIDGROUPID='-1' OUTBOUND CID SELECTED IS '0'.
-- AGI Script Executing Application: (Dial) Options: (SIP/vbuzzer.com/14168805266|60|HRgrL(11850000:61000:30000))
-- Limit Data for this call:
> timelimit = 11850000
> play_warning = 61000
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 30000
> start_sound = (null)
> warning_sound = timeleft
> end_sound = (null)
-- Called vbuzzer.com/14168805266
-- SIP/vbuzzer.com-0a0a3548 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
a2billing.php|1: file:Class.RateEngine.php - line:1079 - [K=0]:[ANSWEREDTIME=0-DIALSTATUS=CONGESTION]
a2billing.php|1: file:Class.RateEngine.php - line:1119 - FAILOVER app_callingcard: Dialing 'SIP/vbuzzer.com/14168805266|60|HRgrL(11850000:61000:30000)' with timeout of '11850'.
a2billing.php|1:
-- AGI Script Executing Application: (DIAL) Options: (SIP/vbuzzer.com/14168805266|60|HRgrL(11850000:61000:30000))
-- Limit Data for this call:
> timelimit = 11850000
> play_warning = 61000
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 30000
> start_sound = (null)
> warning_sound = timeleft
> end_sound = (null)
-- Called vbuzzer.com/14168805266
-- SIP/vbuzzer.com-0a0a3548 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
a2billing.php|1: file:Class.RateEngine.php - line:1129 - [FAILOVER K=0]:[ANSTIME=-DIALSTATUS=CONGESTION]
a2billing.php|1: file:Class.RateEngine.php - line:1079 - [K=0]:[ANSWEREDTIME=0-DIALSTATUS=CONGESTION]
a2billing.php|1: file:Class.RateEngine.php - line:1119 - FAILOVER app_callingcard: Dialing 'SIP/vbuzzer.com/14168805266|60|HRgrL(11850000:61000:30000)' with timeout of '11850'.
a2billing.php|1:
-- AGI Script Executing Application: (DIAL) Options: (SIP/vbuzzer.com/14168805266|60|HRgrL(11850000:61000:30000))
-- Limit Data for this call:
> timelimit = 11850000
> play_warning = 61000
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 30000
> start_sound = (null)
> warning_sound = timeleft
> end_sound = (null)
-- Called vbuzzer.com/14168805266
-- SIP/vbuzzer.com-0a0a3548 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
a2billing.php|1: file:Class.RateEngine.php - line:1129 - [FAILOVER K=0]:[ANSTIME=-DIALSTATUS=CONGESTION]
a2billing.php|1: file:Class.RateEngine.php - line:1079 - [K=0]:[ANSWEREDTIME=0-DIALSTATUS=CONGESTION]
a2billing.php|1: file:Class.RateEngine.php - line:1119 - FAILOVER app_callingcard: Dialing 'SIP/vbuzzer.com/14168805266|60|HRgrL(11850000:61000:30000)' with timeout of '11850'.
a2billing.php|1:
-- AGI Script Executing Application: (DIAL) Options: (SIP/vbuzzer.com/14168805266|60|HRgrL(11850000:61000:30000))
-- Limit Data for this call:
> timelimit = 11850000
> play_warning = 61000
> play_to_caller = yes
> play_to_callee = no
> warning_freq = 30000
> start_sound = (null)
> warning_sound = timeleft
> end_sound = (null)
-- Called vbuzzer.com/14168805266
-- SIP/vbuzzer.com-0a0a3548 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
a2billing.php|1: file:Class.RateEngine.php - line:1129 - [FAILOVER K=0]:[ANSTIME=-DIALSTATUS=CONGESTION]
a2billing.php|1: file:Class.RateEngine.php - line:1162 - [USEDRATECARD - FAIL =0]
-- Playing 'prepaid-dest-unreachable' (escape_digits=#) (sample_offset 0)
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
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 ('1211411947.5', 'SIP/felixjoe-0a09dc58', '5388980051', '', CURRENT_TIMESTAMP - INTERVAL 0 SECOND , '0', '14168805266', 'CONGESTION', now(), '0.02', '-0', '', '', 'ca', '3', '3', '3', '5', '9057921343', '0', '0.01', '0', '0')]
a2billing.php|1: file:Class.RateEngine.php - line:917 - [CC_asterisk_stop 1.1: SQL: DONE : result=1]
a2billing.php|1: file:a2billing.php - line:310 - [a2billing account stop]
a2billing.php|1: file:a2billing.php - line:170 - [CHANNEL STATUS : 6 = Line is up]
a2billing.php|1: file:a2billing.php - line:171 - [CREDIT : 4.15000][CREDIT MIN_CREDIT_2CALL : 0]
a2billing.php|1: file:Class.A2Billing.php - line:671 - 0 && && 8&& 1
-- <SIP/felixjoe-0a09dc58> Playing 'prepaid-enter-dest' (language 'en')
a2billing.php|1: file:Class.A2Billing.php - line:678 - RES DTMF : -1
a2billing.php|1: file:Class.A2Billing.php - line:696 - DESTINATION ::> -1
a2billing.php|1: file:Class.A2Billing.php - line:698 - RULES APPLY ON DESTINATION ::> -1
-- Playing 'prepaid-invalid-digits' (escape_digits=#) (sample_offset 0)
a2billing.php|1: file:a2billing.php - line:170 - [CHANNEL STATUS : 6 = Line is up]
a2billing.php|1: file:a2billing.php - line:171 - [CREDIT : 4.15000][CREDIT MIN_CREDIT_2CALL : 0]
a2billing.php|1: file:Class.A2Billing.php - line:671 - 0 && && 8&& 2
-- <SIP/felixjoe-0a09dc58> Playing 'prepaid-enter-dest' (language 'en')
a2billing.php|1: file:Class.A2Billing.php - line:678 - RES DTMF : -1
a2billing.php|1: file:Class.A2Billing.php - line:696 - DESTINATION ::> -1
a2billing.php|1: file:Class.A2Billing.php - line:698 - RULES APPLY ON DESTINATION ::> -1
-- Playing 'prepaid-invalid-digits' (escape_digits=#) (sample_offset 0)
a2billing.php|1: file:Class.A2Billing.php - line:649 - [CARD STATUS UPDATE : UPDATE cc_card SET inuse=inuse-1 WHERE username='5388980051']
-- AGI Script a2billing.php completed, returning -1
trixbox1*CLI>


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 Post subject:
PostPosted: Thu May 22, 2008 4:48 am 
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Joined: Fri Jun 23, 2006 3:56 pm
Posts: 4065
You do not appear to have followed Stavro's advice:-

Quote:
If that doesn't help, my next guess would be to try renaming your peer definition so it doesn't look quite so much like a valid domain name. Don't forget to amend any associated A2Billing trunk definitions to use the new name in the provider IP field.


Try it, I can see how that may be causing a problem.

Joe


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 Post subject: Re: outbound error
PostPosted: Tue Sep 15, 2009 2:57 pm 
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Joined: Mon Sep 14, 2009 10:45 pm
Posts: 8
I have the same problem. May be anybody could post the solution. I listen one tone and then "The numer you have dialed is unavaiable. Pleasse check te number and try again". I using elastix and my trunk is working fine.


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