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"Number Doesn't answer" ?
http://forum.asterisk2billing.org/viewtopic.php?f=21&t=6692		 Page 1 of 1
Author:  rboy22007 [ Tue Dec 01, 2009 11:35 pm ]
Post subject:  "Number Doesn't answer" ?
hi...

when i try to make call out of a2billing using my sip trunk..i get "number doesn't answer" message straight away.. is it something to do with my trunk setting or the sip provide is incompatible. i am using sipgate and trying to dial into a2billing then dial out after authentication.

the log spit out the following:

[Dec 1 23:20:00] VERBOSE[3443] logger.c: == Manager 'admin' logged off from 127.0.0.1
[Dec 1 23:20:05] VERBOSE[2506] logger.c: == Using SIP RTP TOS bits 184
[Dec 1 23:20:05] VERBOSE[2506] logger.c: == Using SIP RTP CoS mark 5
[Dec 1 23:20:05] VERBOSE[2506] logger.c: == Using SIP VRTP TOS bits 136
[Dec 1 23:20:05] VERBOSE[2506] logger.c: == Using SIP VRTP CoS mark 6
[Dec 1 23:20:05] VERBOSE[3451] logger.c: -- Executing [4119417@ext-did:1] Set("SIP/4119417-09709a90", "__FROM_DID=4119417") in new stack
[Dec 1 23:20:05] VERBOSE[3451] logger.c: -- Executing [4119417@ext-did:2] Gosub("SIP/4119417-09709a90", "app-blacklist-check,s,1") in new stack
[Dec 1 23:20:05] VERBOSE[3451] logger.c: -- Executing [s@app-blacklist-check:1] GotoIf("SIP/4119417-09709a90", "0?blacklisted") in new stack
[Dec 1 23:20:05] VERBOSE[3451] logger.c: -- Executing [s@app-blacklist-check:2] Return("SIP/4119417-09709a90", "") in new stack
[Dec 1 23:20:05] VERBOSE[3451] logger.c: -- Executing [4119417@ext-did:3] ExecIf("SIP/4119417-09709a90", "0 ?Set(CALLERID(name)=02084712971)") in new stack
[Dec 1 23:20:05] VERBOSE[3451] logger.c: -- Executing [4119417@ext-did:4] Set("SIP/4119417-09709a90", "__CALLINGPRES_SV=allowed_not_screened") in new stack
[Dec 1 23:20:05] VERBOSE[3451] logger.c: -- Executing [4119417@ext-did:5] Set("SIP/4119417-09709a90", "CALLERPRES()=allowed_not_screened") in new stack
[Dec 1 23:20:05] VERBOSE[3451] logger.c: -- Executing [4119417@ext-did:6] Goto("SIP/4119417-09709a90", "custom-a2billing,4119417,1") in new stack
[Dec 1 23:20:05] VERBOSE[3451] logger.c: -- Goto (custom-a2billing,4119417,1)
[Dec 1 23:20:05] VERBOSE[3451] logger.c: -- Executing [4119417@custom-a2billing:1] Answer("SIP/4119417-09709a90", "") in new stack
[Dec 1 23:20:05] VERBOSE[3451] logger.c: -- Executing [4119417@custom-a2billing:2] Wait("SIP/4119417-09709a90", "1") in new stack
[Dec 1 23:20:05] NOTICE[3451] channel.c: Dropping incompatible voice frame on SIP/4119417-09709a90 of format ulaw since our native format has changed to 0x8 (alaw)
[Dec 1 23:20:06] VERBOSE[3451] logger.c: -- Executing [4119417@custom-a2billing:3] DeadAGI("SIP/4119417-09709a90", "a2billing.php,1") in new stack
[Dec 1 23:20:06] WARNING[3451] res_agi.c: DeadAGI has been deprecated, please use AGI in all cases!
[Dec 1 23:20:06] VERBOSE[3451] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
[Dec 1 23:20:06] VERBOSE[3453] logger.c: == Manager 'admin' logged on from 127.0.0.1
[Dec 1 23:20:07] VERBOSE[3451] logger.c: -- <SIP/4119417-09709a90> Playing 'prepaid-enter-pin-number.gsm' (language 'en')
[Dec 1 23:20:08] VERBOSE[3453] logger.c: == Manager 'admin' logged off from 127.0.0.1
[Dec 1 23:20:14] VERBOSE[3463] logger.c: == Manager 'admin' logged on from 127.0.0.1
[Dec 1 23:20:16] VERBOSE[3463] logger.c: == Manager 'admin' logged off from 127.0.0.1
[Dec 1 23:20:16] VERBOSE[3471] logger.c: == Manager 'admin' logged on from 127.0.0.1
[Dec 1 23:20:16] VERBOSE[2506] logger.c: -- ast_get_srv: SRV lookup for '_sip._UDP.sipgate.co.uk' mapped to host sipgate.co.uk, port 5060
[Dec 1 23:20:16] VERBOSE[3471] logger.c: == Manager 'admin' logged off from 127.0.0.1
[Dec 1 23:20:22] VERBOSE[3474] logger.c: == Manager 'admin' logged on from 127.0.0.1
[Dec 1 23:20:23] VERBOSE[3474] logger.c: == Manager 'admin' logged off from 127.0.0.1
[Dec 1 23:20:24] VERBOSE[3451] logger.c: -- Playing 'prepaid-you-have' (escape_digits=#) (sample_offset 0)
[Dec 1 23:20:26] VERBOSE[3451] logger.c: -- <SIP/4119417-09709a90> Playing 'digits/5.gsm' (language 'en')
[Dec 1 23:20:26] NOTICE[3451] rtp.c: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 217.10.69.13
[Dec 1 23:20:26] VERBOSE[3451] logger.c: -- Playing 'credit' (escape_digits=#) (sample_offset 0)
[Dec 1 23:20:27] VERBOSE[3451] logger.c: -- Playing 'vm-and' (escape_digits=#) (sample_offset 0)
[Dec 1 23:20:28] VERBOSE[3451] logger.c: -- <SIP/4119417-09709a90> Playing 'digits/70.gsm' (language 'en')
[Dec 1 23:20:29] VERBOSE[3451] logger.c: -- <SIP/4119417-09709a90> Playing 'digits/6.gsm' (language 'en')
[Dec 1 23:20:29] VERBOSE[3482] logger.c: == Manager 'admin' logged on from 127.0.0.1
[Dec 1 23:20:29] VERBOSE[3451] logger.c: -- Playing 'prepaid-cents' (escape_digits=#) (sample_offset 0)
[Dec 1 23:20:30] VERBOSE[3451] logger.c: -- <SIP/4119417-09709a90> Playing 'prepaid-enter-dest.gsm' (language 'en')
[Dec 1 23:20:31] VERBOSE[3482] logger.c: == Manager 'admin' logged off from 127.0.0.1
[Dec 1 23:20:37] VERBOSE[3490] logger.c: == Manager 'admin' logged on from 127.0.0.1
[Dec 1 23:20:38] VERBOSE[3490] logger.c: == Manager 'admin' logged off from 127.0.0.1
[Dec 1 23:20:41] VERBOSE[3451] logger.c: -- Playing 'prepaid-you-have' (escape_digits=#) (sample_offset 0)
[Dec 1 23:20:42] VERBOSE[3451] logger.c: -- <SIP/4119417-09709a90> Playing 'digits/1.gsm' (language 'en')
[Dec 1 23:20:43] VERBOSE[3451] logger.c: -- <SIP/4119417-09709a90> Playing 'digits/hundred.gsm' (language 'en')
[Dec 1 23:20:44] VERBOSE[3451] logger.c: -- Playing 'prepaid-minutes' (escape_digits=#) (sample_offset 0)
[Dec 1 23:20:44] VERBOSE[3498] logger.c: == Manager 'admin' logged on from 127.0.0.1
[Dec 1 23:20:45] VERBOSE[3451] logger.c: -- AGI Script Executing Application: (DIAL) Options: (SIP/SIPGATE/07904164760|60|HRrL(6000000:61000:30000))
[Dec 1 23:20:45] VERBOSE[3451] logger.c: == Using SIP RTP TOS bits 184
[Dec 1 23:20:45] VERBOSE[3451] logger.c: == Using SIP RTP CoS mark 5
[Dec 1 23:20:45] VERBOSE[3451] logger.c: == Using SIP VRTP TOS bits 136
[Dec 1 23:20:45] VERBOSE[3451] logger.c: == Using SIP VRTP CoS mark 6
[Dec 1 23:20:45] VERBOSE[3451] logger.c: -- Called SIPGATE/07904164760|60|HRrL(6000000:61000:30000)
[Dec 1 23:20:45] VERBOSE[2506] logger.c: -- Got SIP response 475 "Bad URI (475/SL)" back from 217.10.79.23
[Dec 1 23:20:45] VERBOSE[3451] logger.c: -- No one is available to answer at this time (1:0/0/0)
[Dec 1 23:20:45] VERBOSE[3451] logger.c: -- Playing 'prepaid-noanswer' (escape_digits=#) (sample_offset 0)
[Dec 1 23:20:46] VERBOSE[3498] logger.c: == Manager 'admin' logged off from 127.0.0.1
[Dec 1 23:20:46] VERBOSE[3506] logger.c: == Manager 'admin' logged on from 127.0.0.1
[Dec 1 23:20:47] VERBOSE[3506] logger.c: == Manager 'admin' logged off from 127.0.0.1
[Dec 1 23:20:47] VERBOSE[3451] logger.c: -- <SIP/4119417-09709a90> Playing 'prepaid-enter-dest.gsm' (language 'en')
[Dec 1 23:20:50] VERBOSE[3451] logger.c: -- <SIP/4119417-09709a90>AGI Script a2billing.php completed, returning -1

much appreciate your help...
Author:  jroper [ Wed Dec 02, 2009 12:29 am ]
Post subject:  Re: "Number Doesn't answer" ?
Quote:
[Dec 1 23:20:45] VERBOSE[3451] logger.c: -- Called SIPGATE/07904164760|60|HRrL(6000000:61000:30000)
[Dec 1 23:20:45] VERBOSE[2506] logger.c: -- Got SIP response 475 "Bad URI (475/SL)" back from 217.10.79.23


You may get more joy if you sent 447904164760 to Sipgate.

Joe
Author:  rboy22007 [ Thu Dec 03, 2009 1:14 pm ]
Post subject:  Re: "Number Doesn't answer" ?
hi..thanks for the replay..

i have tried as you said with the country code 44, 044,

i didn't work..it keep saying "the number is not answering"

i still get 475 Bad URI.

[Dec 3 13:00:25] VERBOSE[3180] logger.c: -- Called VOIPTALK/442084712971|60|HRrL(12000000:61000:30000)
[Dec 3 13:00:25] VERBOSE[2516] logger.c: -- Got SIP response 475 "Bad URI (475/SL)" back from 77.240.48.94
[Dec 3 13:00:25] VERBOSE[3180] logger.c: -- No one is available to answer at this time (1:0/0/0)
[Dec 3 13:00:25] VERBOSE[3180] logger.c: -- Playing 'prepaid-noanswer' (escape_digits=#) (sample_offset 0)

i use sipgate to Receive the incoming call to A2BILLING and dial Out using Voiptalk:

i got the following trunk setting for voiptalk;

register=> 844246509:[email protected]/844246509
type=friend
username=844246509
secret=xxxxxx
fromuser=844246509
host=voiptalk.org
dtmfmode=rfc2833
fromdomain=voiptalk.org
context=default
insecure=very

https://www.voiptalk.org/products/aster ... 00bd7d6d3d
Author:  rboy22007 [ Tue Dec 08, 2009 4:14 pm ]
Post subject:  The numer is currently unavailable
Hi...

i set up an IAX2 trunk with VoipTlak
and now iam getting..Number is currently Unavailable

i get the following error:

[Dec 8 16:04:32] VERBOSE[2421] logger.c: --- (9 headers 0 lines) ---
[Dec 8 16:04:32] VERBOSE[2421] logger.c:
<--- SIP read from UDP://77.240.48.94:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK29d4dd35;rport=5060
From: <sip:[email protected]>;tag=as1c47720b
To: <sip:[email protected]>;tag=fd79486175647ed1617969929fdaad02.b0b3
Call-ID: [email protected]
CSeq: 109 REGISTER
Contact: <sip:[email protected]>;expires=120
Server: OpenSIPS (1.5.3-notls (x86_64/linux))
Content-Length: 0
Warning: 392 77.240.48.94:5060 "Noisy feedback tells: pid=26507 req_src_ip=79.78.21.93 req_src_port=5060 in_uri=sip:voiptalk.org out_uri=sip:voiptalk.org via_cnt==1"


<------------->
[Dec 8 16:04:32] VERBOSE[2421] logger.c: --- (10 headers 0 lines) ---
[Dec 8 16:04:32] VERBOSE[2421] logger.c: Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
[Dec 8 16:04:32] NOTICE[2421] chan_sip.c: Outbound Registration: Expiry for voiptalk.org is 120 sec (Scheduling reregistration in 105 s)
[Dec 8 16:04:36] VERBOSE[5819] logger.c: -- Playing 'prepaid-you-have' (escape_digits=#) (sample_offset 0)
[Dec 8 16:04:37] VERBOSE[5819] logger.c: -- <SIP/4119417-b6e16e00> Playing 'digits/5.gsm' (language 'en')
[Dec 8 16:04:38] NOTICE[5819] rtp.c: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 217.10.69.13
[Dec 8 16:04:38] VERBOSE[5819] logger.c: -- Playing 'credit' (escape_digits=#) (sample_offset 0)
[Dec 8 16:04:39] VERBOSE[5819] logger.c: -- Playing 'vm-and' (escape_digits=#) (sample_offset 0)
[Dec 8 16:04:39] VERBOSE[5819] logger.c: -- <SIP/4119417-b6e16e00> Playing 'digits/70.gsm' (language 'en')
[Dec 8 16:04:40] VERBOSE[5819] logger.c: -- <SIP/4119417-b6e16e00> Playing 'digits/6.gsm' (language 'en')
[Dec 8 16:04:41] VERBOSE[5819] logger.c: -- Playing 'prepaid-cents' (escape_digits=#) (sample_offset 0)
[Dec 8 16:04:42] VERBOSE[5819] logger.c: -- <SIP/4119417-b6e16e00> Playing 'prepaid-enter-dest.gsm' (language 'en')
[Dec 8 16:04:56] VERBOSE[5819] logger.c: -- Playing 'prepaid-you-have' (escape_digits=#) (sample_offset 0)
[Dec 8 16:04:57] VERBOSE[5819] logger.c: -- <SIP/4119417-b6e16e00> Playing 'digits/1.gsm' (language 'en')
[Dec 8 16:04:58] VERBOSE[5819] logger.c: -- <SIP/4119417-b6e16e00> Playing 'digits/hundred.gsm' (language 'en')
[Dec 8 16:04:59] VERBOSE[5819] logger.c: -- Playing 'prepaid-minutes' (escape_digits=#) (sample_offset 0)
[Dec 8 16:05:00] VERBOSE[5819] logger.c: -- AGI Script Executing Application: (DIAL) Options: (IAX2/IAX2/442084712971|60|HRrL(6000000:61000:30000))
[Dec 8 16:05:00] DEBUG[5819] chan_iax2.c: prepending 8 to prefs
[Dec 8 16:05:00] VERBOSE[5819] logger.c: -- Called IAX2/442084712971|60|HRrL(6000000:61000:30000)
[Dec 8 16:05:00] WARNING[2458] chan_iax2.c: Call rejected by 217.14.138.130: No authority found
[Dec 8 16:05:00] VERBOSE[5819] logger.c: -- Hungup 'IAX2/IAX2-2705'
[Dec 8 16:05:00] VERBOSE[5819] logger.c: == Everyone is busy/congested at this time (1:0/0/1)
[Dec 8 16:05:00] VERBOSE[5819] logger.c: -- Playing 'prepaid-dest-unreachable' (escape_digits=#) (sample_offset 0)
[Dec 8 16:05:03] VERBOSE[5819] logger.c: -- <SIP/4119417-b6e16e00> Playing 'prepaid-enter-dest.gsm' (language 'en')
[Dec 8 16:05:04] VERBOSE[2421] logger.c: Really destroying SIP dialog '[email protected]' Method: REGISTER
[Dec 8 16:05:04] VERBOSE[2421] logger.c: Really destroying SIP dialog '[email protected]' Method: REGISTER
[Dec 8 16:05:14] VERBOSE[5819] logger.c: -- Playing 'prepaid-you-have' (escape_digits=#) (sample_offset 0)
[Dec 8 16:05:15] VERBOSE[5819] logger.c: -- <SIP/4119417-b6e16e00> Playing 'digits/1.gsm' (language 'en')
[Dec 8 16:05:16] VERBOSE[5819] logger.c: -- <SIP/4119417-b6e16e00> Playing 'digits/hundred.gsm' (language 'en')
[Dec 8 16:05:17] VERBOSE[5819] logger.c: -- Playing 'prepaid-minutes' (escape_digits=#) (sample_offset 0)
[Dec 8 16:05:17] VERBOSE[2421] logger.c: Reliably Transmitting (no NAT) to 217.10.79.23:5060:
OPTIONS sip:sipgate.co.uk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.4:5060;branch=z9hG4bK236f53c8;rport
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as1054ce1e
To: <sip:sipgate.co.uk>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40
Date: Tue, 08 Dec 2009 16:05:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

any Idea? is it A2Billing not sending in right format to the Provider?

help........ :|
Author:  jroper [ Tue Dec 08, 2009 5:13 pm ]
Post subject:  Re: "Number Doesn't answer" ?
Hi

Shouldn't you be sending 0844246509 or 44844246509?

Joe
Author:  rboy22007 [ Sun Dec 13, 2009 10:25 pm ]
Post subject:  Re: "Number Doesn't answer" ?
hi..i tried sending with 44 or with out 44..its no use. the trunk is working fine outside a2billing.
i've been searching the forum but could find any solution to this problem. i done a debug i am keep getting "Everyone is busy/congested at this time". i've attached the log. could anyone kindly have a look and put me out of this misery. i've been working on this for nearly a month with no solution to this problem. i am suspecting that a2billing in not hitting the trunk in pbx. you probably notice i got ip address of the provider in the a2billing trunk, when i put name as it set up in pbx i get message straightaway "the number doesn't answer" but when i put the ip address, it take some time then i get "the number is not available" message.

do i have to add something in the extension_custom so that a2billing could route call out?

i got the following in

; This file contains example extensions_custom.conf entries.
; extensions_custom.conf should be used to include customizations
; to AMP's Asterisk dialplan.

; All custom context should contain the string 'custom' in it's name

; Extensions in AMP have access to the 'from-internal' context.
; The context 'from-internal-custom' is included in 'from-internal' by default

#include extensions_hud.conf

[from-internal-custom]

;1234,1,Playback(demo-congrats) ; extensions can dial 1234
;1234,2,Hangup()
;h,1,Hangup()
;include => custom-recordme ; extensions can also dial 5678

; custom-count2four,s,1 can be used as a custom target for
; a Digital Receptionist menu or a Call Group
;[custom-count2four]
;s,1,SayDigits(1234)
;s,2,Hangup

; custom-recordme,5678,1 can be used as a custom target for
; a Digital Receptionist menu or a Call Group
;[custom-recordme]
;exten => 5678,1,Wait(2)
;exten => 5678,2,Record(/tmp/asterisk-recording:gsm)
;exten => 5678,3,Wait(2)
;exten => 5678,4,Playback(/tmp/asterisk-recording)
;exten => 5678,5,Wait(2)
;exten => 5678,6,Hangup

[custom-meetme3]
exten => s,1,Answer
exten => s,n,Wait(3)
exten => s,n,CBMysql()
exten => s,n,Hangup

[a2billing]
exten => _X.,1,Answer
exten => _X.,n,Wait(1)
exten => _X.,n,deadAGI(a2billing.php,1)
exten => _X.,n,Hangup

[custom-a2billing]
exten => _X.,1,Answer
exten => _X.,n,Wait(1)
exten => _X.,n,deadAGI(a2billing.php,1)
exten => _X.,n,Hangup


[macro-dialout-trunk-predial-hook]
exten => s,1,GotoIf($["${OUT_${DIAL_TRUNK}:4:4}" = "A2B/"]?custom-freepbx-a2billing,${OUTNUM},1:2)
exten => s,2,MacroExit

[custom-freepbx-a2billing]
exten => _X.,1,DeadAGI(a2billing.php,1|${OUT_${DIAL_TRUNK}:8})
exten => _X.,n,Hangup()
Author:  jroper [ Mon Dec 14, 2009 7:52 am ]
Post subject:  Re: "Number Doesn't answer" ?
In provider IP in A2Billing, change "217.14.138.10" to VoipTalk as per this instructions against that field.

Quote:
Set the IP or URL of the VoIP provider. Alternatively, put in the name of a previously defined trunk in Asterisk or FreePBX.
Author:  rboy22007 [ Sat Dec 19, 2009 10:08 pm ]
Post subject:  Re: "Number Doesn't answer" ?
hi..

i have tried putting as Voiptalk (the same name i got in the trixbox trunk)
how can i check if the a2b is passing the call to tb call out dial plan?
Author:  rboy22007 [ Mon Dec 21, 2009 11:42 am ]
Post subject:  Re: "Number Doesn't answer" ? (SOLVED)
hi...

never mind, i've solved the problem. it required some tweak in agi-config.
thanks jroper for your advice n help. :mrgreen2:
Author:  taphawane [ Sat Dec 11, 2010 8:16 am ]
Post subject:  Re: "Number Doesn't answer" ?
Hello,

I have a same problem, you said you have resolved can you please share with us
Author:  rboy22007 [ Sat Dec 11, 2010 11:18 am ]
Post subject:  Re: "Number Doesn't answer" ?
Certainly, I'l share it with you.

First of all it depends on what version of asterisk you are using, i had the problem in asterisk 1.6, where in agi- config i had to change the Dial Command Params from this (|60|HRL(%timeout%:61000:30000) to this (,60|HRL(%timeout%:61000:30000) and you have to match this in your custom extension in your PBX.

That's how i've solved my problem but it would help if you could attach your log file, i'm not an expert but i'm sure someone in this wonderfull forum would help you.
Author:  sandman002 [ Fri Sep 09, 2011 3:05 am ]
Post subject:  Re: "Number Doesn't answer" ?
Thanks rboy finding this thread solved my problem of unsupported URI. Been racking my head against this one for a week now, the pipe change did the trick. I had opened another thread looking for help with this.

Sandman
Author:  jroper [ Fri Sep 09, 2011 6:16 am ]
Post subject:  Re: "Number Doesn't answer" ?
Setting the asterisk version to 1_6 in the Global and agi-conf would have worked as well.
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