OUTBOUND call not working .

[heart200306]

Dear Guys

would you please help me to outbound issue . when I am trying to make manual call from agent web panel I am finding waiting for ring after 60 sec is auto dc but call is not hit carrier gw or callee(destnation) number .

Dialed number :447828333980
useing Dial System :9+ then Number


Asterisk 1.4.44-vici

ISO ::ViciBox.x86_64-4.0.3.iso
ViciBox Server v.4.0!
SVN Version: 1953
DB Schema Version: 1341
Kernel:
Linux ccdial1 3.1.10-1.19-default #1 SMP Mon Feb 25 10:32:50 UTC 2013 (f0b13a3) x86_64 x86_64 x86_64 GNU/Linux
VERSION: 2.6-396a
BUILD: 130402-2322

Server :Cluster:: No senguma , No other SW install .

::::extensions.conf

exten => 8365,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8365,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----SO)
exten => 8365,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----SO)
exten => 8365,5,Hangup

; VICIDIAL_auto_dialer transfer script SURVEY at beginning:
exten => 8366,1,Playback(sip-silence)
exten => 8366,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8366,3,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)
exten => 8366,4,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)
exten => 8366,5,Hangup

; VICIDIAL_auto_dialer transfer script Load Balance Overflow:
exten => 8367,1,Playback(sip-silence)
exten => 8367,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8367,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LO)
exten => 8367,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LO)
exten => 8367,5,Hangup

; VICIDIAL_auto_dialer transfer script Load Balanced:
exten => 8368,1,Playback(sip-silence)
exten => 8368,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8368,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)


Dial Plan :

exten => _944X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _944X.,n,Dial(SIP/out1/${EXTEN:1},,To)
exten => _944X.,n,HangUp

Astericks Log :

<------------->
[Apr 21 05:05:58] == Parsing '/etc/asterisk/manager.conf': [Apr 21 05:05:58] Found
[Apr 21 05:05:58] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 21 05:05:58] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-ef2b,2", "8600051|F") in new stack
[Apr 21 05:05:58] == Parsing '/etc/asterisk/meetme.conf': [Apr 21 05:05:58] Found
[Apr 21 05:05:58] == Parsing '/etc/asterisk/meetme-vicidial.conf': [Apr 21 05:05:58] Found
[Apr 21 05:05:58] > Channel Local/8600051@default-ef2b,1 was answered.
[Apr 21 05:05:58] == Starting Local/8600051@default-ef2b,1 at default,99447828333980,1 failed so falling back to exten 's'
[Apr 21 05:05:58] == Starting Local/8600051@default-ef2b,1 at default,s,1 still failed so falling back to context 'default'
[Apr 21 05:05:58] -- Sent into invalid extension 's' in context 'default' on Local/8600051@default-ef2b,1
[Apr 21 05:05:58] -- Created MeetMe conference 1023 for conference '8600051'
[Apr 21 05:05:58] -- Executing [i@default:1] Playback("Local/8600051@default-ef2b,1", "invalid") in new stack
[Apr 21 05:05:58] -- <Local/8600051@default-ef2b,2> Playing 'conf-onlyperson' (language 'en')
[Apr 21 05:05:58] -- <Local/8600051@default-ef2b,1> Playing 'invalid' (language 'en')
[Apr 21 05:05:58] WARNING[15608]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
[Apr 21 05:06:00] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 21 05:06:01] WARNING[15608]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
[Apr 21 05:06:01] == Parsing '/etc/asterisk/manager.conf': [Apr 21 05:06:01] Found
[Apr 21 05:06:01] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 21 05:06:01] == Parsing '/etc/asterisk/manager.conf': [Apr 21 05:06:01] Found
[Apr 21 05:06:01] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 21 05:06:01] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 21 05:06:02] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 21 05:06:02] == Auto fallthrough, channel 'Local/8600051@default-ef2b,1' status is 'UNKNOWN'
[Apr 21 05:06:02] -- Executing [h@default:1] DeadAGI("Local/8600051@default-ef2b,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr 21 05:06:02] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 21 05:06:02] -- Hungup 'DAHDI/pseudo-623666775'
[Apr 21 05:06:02] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-ef2b,2'
[Apr 21 05:06:02] -- Executing [h@default:1] DeadAGI("Local/8600051@default-ef2b,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr 21 05:06:02] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 21 05:06:06] Scheduling destruction of SIP dialog '29efcfd465b04a20160b266b0e2cfb4b@192.168.1.1' in 19328 ms (Method: INVITE)
[Apr 21 05:06:06] Reliably Transmitting (NAT) to 103.4.144.158:55380:
CANCEL sip:201@103.4.144.158:55124;rinstance=44643664566f64d5;transport=UDP;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK576df3c7;rport
From: "S1304210905368600051" <sip:0000000000@192.168.1.1>;tag=as32c44bd0
To: <sip:201@103.4.144.158:55124;rinstance=44643664566f64d5;transport=UDP;cpd=on>
Call-ID: 29efcfd465b04a20160b266b0e2cfb4b@192.168.1.1
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "S1304210905368600051" <sip:0000000000@192.168.1.1>;privacy=off;screen=no
Content-Length: 0

[Michael_N]

in campaign detail view
change Dial Prefix to X

[heart200306]

Dear Michael
Thank you for quick reply . I had modified Campaign detils and Dial prefix to X but result is same .
I wanna using dial prefix 9 . So I did keep same dial prefix 9.
Now I am dialing 9447828333980 but call is not going through ....

Aster log :

---
[Apr 21 06:12:51] NOTICE[2124]: chan_sip.c:13705 handle_response_invite: Failed to authenticate on INVITE to '"sip201" <sip:201@192.168.1.1>;tag=as11e27c94'
[Apr 21 06:12:51] -- SIP/out1-000000b8 is circuit-busy
[Apr 21 06:12:51] == Everyone is busy/congested at this time (1:0/1/0)
[Apr 21 06:12:51] -- Executing [9447828333980@default:3] Hangup("SIP/201-000000b7", "") in new stack
[Apr 21 06:12:51] == Spawn extension (default, 9447828333980, 3) exited non-zero on 'SIP/201-000000b7'
[Apr 21 06:12:51] -- Executing [h@default:1] DeadAGI("SIP/201-000000b7", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----CONGESTION----------") in new stack
[Apr 21 06:12:51] Really destroying SIP dialog '03c1e99d5e69d48a0966ce9071e7dab3@192.168.1.1' Method: INVITE
[Apr 21 06:12:51] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 21 06:12:51] Scheduling destruction of SIP dialog 'NWYxYTI4NzQ1NDA1ZjNlZDVhYjcwYTA1NGQ2YTI5YTA.' in 32000 ms (Method: INVITE)
[Apr 21 06:12:51]
<--- Reliably Transmitting (NAT) to 103.4.144.158:5061 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 103.4.144.158:5061;branch=z9hG4bK-d8754z-aaaacb5eed19c1d3-1---d8754z-;received=103.4.144.158
From: "123"<sip:201@192.168.1.1;transport=UDP>;tag=1f6f2c46
To: <sip:9447828333980@192.168.1.1;transport=UDP>;tag=as19e3b9cf
Call-ID: NWYxYTI4NzQ1NDA1ZjNlZDVhYjcwYTA1NGQ2YTI5YTA.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
[Apr 21 06:12:51]
<--- SIP read from 103.4.144.158:5061 --->



<------------->
[Apr 21 06:12:52]
<--- SIP read from 103.4.144.158:5061 --->
ACK sip:9447828333980@192.168.1.1;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 103.4.144.158:5061;branch=z9hG4bK-d8754z-aaaacb5eed19c1d3-1---d8754z-
Max-Forwards: 70
To: <sip:9447828333980@192.168.1.1;transport=UDP>;tag=as19e3b9cf
From: "123"<sip:201@192.168.1.1;transport=UDP>;tag=1f6f2c46
Call-ID: NWYxYTI4NzQ1NDA1ZjNlZDVhYjcwYTA1NGQ2YTI5YTA.
CSeq: 2 ACK
Content-Length: 0


<------------->

[heart200306]

Anyone can help me please ...

[williamconley]

i don't see the "dial" command in either of your examples. you are starting your CLI output too late. we need to see the entire process. STARTING with the error does not help us to understand what happened to cause the error. show us all output from the dial attempt. (good that you kept output to a minimum, nothing unrelated ... but we do need from the inception of the call).

[heart200306]

Thanks william .I had find my problem which is load list problem. Anyway .. would you please send lead loader list sample csv file to my email . heart200306@yahoo.com