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 Post subject: Callback hangup after 20 seconds
PostPosted: Tue Jan 22, 2008 9:02 pm 
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Joined: Sat Mar 31, 2007 10:30 pm
Posts: 28
Location: USA
Hi Guys,
Please I am having an issue with my callback, I am using CID-Callback, when you call the access number, it calls back and ask for the destination number, as you enter the destination number the call will hangup about 20 - 24 seconds, as you are entering the destination number. Can anyone tell me what is the problem, if you think it's a2b, asterisk or carrier problem, but if it's the carrier, which carrier is good for callback service.

Regards,
Layer3.


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 Post subject:
PostPosted: Tue Jan 22, 2008 9:12 pm 
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Joined: Wed Sep 12, 2007 5:31 pm
Posts: 21
same thing is happening with me....i can't figure it out but sometimes it works ok....and sometime it just says the numb er you are dialing is not available and it hungs up approx 30 sec


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 Post subject:
PostPosted: Tue Jan 22, 2008 9:23 pm 
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Joined: Tue Jan 22, 2008 8:42 pm
Posts: 9
same this for me also pls help me someone

Phone Callback
a2billing.php|1|all-callback|1: file:Class.A2Billing.php - line:616 -
get_agi_request_parameter = 4167544616 ; SIP/1207878261-b7506510 ;
1201032943.18 ; ; 6474766001
a2billing.php|1|all-callback|1: file:a2billing.php - line:555 - [MODE :
ALL-CALLBACK - 4167544616]
a2billing.php|1|all-callback|1: file:a2billing.php - line:559 - [HANGUP
ALL CALLBACK TRIGGER]
== Spawn extension (voicenetwork, 6474766001, 1) exited non-zero on
'SIP/1207878261-b7506510'
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'myasterisk' logged on from 127.0.0.1
== Starting SIP/a2billing-callback-08e9d528 at a2billing-callback,1000,1
failed so falling back to exten 's'
== Starting SIP/a2billing-callback-08e9d528 at a2billing-callback,s,1
still failed so falling back to context 'default'
-- Executing [s@default:1] Playback("SIP/a2billing-callback-08e9d528",
"vm-goodbye") in new stack
-- <SIP/a2billing-callback-08e9d528> Playing 'vm-goodbye' (language
'en')
== Manager 'myasterisk' logged off from 127.0.0.1
-- Executing [s@default:2] Macro("SIP/a2billing-callback-08e9d528",
"hangupcall") in new stack
-- Executing [s@macro-hangupcall:1]
ResetCDR("SIP/a2billing-callback-08e9d528", "w") in new stack
-- Executing [s@macro-hangupcall:2]
NoCDR("SIP/a2billing-callback-08e9d528", "") in new stack
-- Executing [s@macro-hangupcall:3]
GotoIf("SIP/a2billing-callback-08e9d528", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6]
GotoIf("SIP/a2billing-callback-08e9d528", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9]
GotoIf("SIP/a2billing-callback-08e9d528", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11]
Hangup("SIP/a2billing-callback-08e9d528", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/a2billing-callback-08e9d528' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/a2billing-callback-08e9d528'
elastix*CLI>

Web callback
== Manager 'myasterisk' logged on from 127.0.0.1
== Starting SIP/a2billing-callback-08e9d528 at
a2billing-callback,14163004463,1 failed so falling back to exten 's'
== Starting SIP/a2billing-callback-08e9d528 at a2billing-callback,s,1
still failed so falling back to context 'default'
-- Executing [s@default:1] Playback("SIP/a2billing-callback-08e9d528",
"vm-goodbye") in new stack
-- <SIP/a2billing-callback-08e9d528> Playing 'vm-goodbye' (language
'en')
== Manager 'myasterisk' logged off from 127.0.0.1
-- Executing [s@default:2] Macro("SIP/a2billing-callback-08e9d528",
"hangupcall") in new stack
-- Executing [s@macro-hangupcall:1]
ResetCDR("SIP/a2billing-callback-08e9d528", "w") in new stack
-- Executing [s@macro-hangupcall:2]
NoCDR("SIP/a2billing-callback-08e9d528", "") in new stack
-- Executing [s@macro-hangupcall:3]
GotoIf("SIP/a2billing-callback-08e9d528", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6]
GotoIf("SIP/a2billing-callback-08e9d528", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9]
GotoIf("SIP/a2billing-callback-08e9d528", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11]
Hangup("SIP/a2billing-callback-08e9d528", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/a2billing-callback-08e9d528' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/a2billing-callback-08e9d528'


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 Post subject:
PostPosted: Tue Jan 22, 2008 9:36 pm 
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Moderator
User avatar

Joined: Thu Jun 22, 2006 2:19 pm
Posts: 2890
Location: Devon, UK
layer3, your problem is likely caused by either your carrier, or Asterisk or a combination of the two. voip-info has a large list of software you might use to help you diagnose this issue. Personally, I'd recommend Wireshark in combination with studying the relevant RFCs. I know there is a problem in some versions of Asterisk 1.2 (and perhaps 1.4 too, I don't use it yet) which can cause symptoms similar to those that you are experiencing with some carriers/endpoints.

Alan, as your problem is intermittent it sounds very much like a carrier issue to me. Again you'll only really know for sure once you look into the issue in detail. Wireshark would be an ideal tool.

Praba, your problem is entirely different. The 'failed so falling back' errors you are getting clearly show you haven't configured your dialplan correctly for A2B callbacks. Try reading the callback installation guide distributed with A2B in the addons/Doc folder.


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 Post subject:
PostPosted: Wed Jan 23, 2008 1:21 pm 
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Joined: Tue Jan 22, 2008 8:42 pm
Posts: 9
which dial plan? and How i even update 1.3.2 still same please help me to solve this issue i when thru all doci cann't find any mistake or where i made please help me

Thanks
praba


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 Post subject:
PostPosted: Mon Oct 13, 2008 7:22 pm 
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Joined: Mon Oct 13, 2008 7:15 pm
Posts: 7
i had/have the same problem with the hangup after around 20-25 sec. - it seems the DTMF tones are not recognized / not sent by the incoming trunk

the problem is A2B have a hardcoded number_try=1 with 6000 msec timeout for all Callbacks - so the simple solution for me is to edit a2billing.php around line 728

$A2B->agiconfig['number_try'] =3;

i like 3 times the question for the customer ... however this not solved the DTMF problem ...

after hrs of trail and error, i found it was a not strict configured incoming trunk, i added these lines to the incoming trunk in the sip.conf

disallow=all
allow=ulaw;alaw
echocancel=yes
dtmfmode=rfc2833

now it is 100% ok - hope this helps ... ;-)

greetings from austria

foxxx


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