vicidial.org

Hi,

I've installed Vicibox 6.0.3 preload successfully and when making call the cli says:-

[Sep 16 01:37:26] == Using SIP RTP CoS mark 5
[Sep 16 01:37:30] > Channel SIP/201-00000008 was answered.
[Sep 16 01:37:30] -- Executing [8600051@default:1] MeetMe("SIP/201-00000008", "8600051,F") in new stack
[Sep 16 01:37:30] == Parsing '/etc/asterisk/meetme.conf': [Sep 16 01:37:30] == Found
[Sep 16 01:37:30] == Parsing '/etc/asterisk/meetme-vicidial.conf': [Sep 16 01:37:30] == Found
[Sep 16 01:37:30] -- Created MeetMe conference 1023 for conference '8600051'
[Sep 16 01:37:30] -- <SIP/201-00000008> Playing 'conf-onlyperson.gsm' (language 'en')
[Sep 16 01:37:31] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 16 01:37:32] NOTICE[27914]: res_rtp_asterisk.c:2382 ast_rtp_read: Unknown RTP codec 126 received from '192.168.1.137:54284'
[Sep 16 01:37:36] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 16 01:37:36] NOTICE[27924]: chan_local.c:973 local_call: No such extension/context 33235580657@default while calling Local channel
[Sep 16 01:37:36] NOTICE[27924]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/33235580657@default
[Sep 16 01:37:36] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 16 01:37:36] NOTICE[27927]: chan_local.c:973 local_call: No such extension/context 33276526400@default while calling Local channel
[Sep 16 01:37:36] NOTICE[27927]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/33276526400@default
[Sep 16 01:37:37] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 16 01:37:37] NOTICE[27930]: chan_local.c:973 local_call: No such extension/context 33276019157@default while calling Local channel
[Sep 16 01:37:37] NOTICE[27930]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/33276019157@default
[Sep 16 01:37:37] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 16 01:37:37] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 16 01:37:38] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 16 01:38:02] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 16 01:38:02] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 16 01:38:02] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 16 01:38:02] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 16 01:38:07] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 16 01:38:07] == Manager 'sendcron' logged off from 127.0.0.1
[Sep 16 01:38:26] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 16 01:38:26] -- Hungup 'DAHDI/pseudo-456519904'
[Sep 16 01:38:26] == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/201-00000008'
[Sep 16 01:38:26] -- Executing [h@default:1] AGI("SIP/201-00000008", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Sep 16 01:38:26] -- <SIP/201-00000008>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Sep 16 01:38:26] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 16 01:38:26] -- Executing [55558600051@default:1] MeetMeAdmin("Local/55558600051@default-00000020;2", "8600051,K") in new stack
[Sep 16 01:38:26] WARNING[28033]: app_meetme.c:4840 admin_exec: Conference number '8600051' not found!
[Sep 16 01:38:26] -- Executing [55558600051@default:2] Hangup("Local/55558600051@default-00000020;2", "") in new stack
[Sep 16 01:38:26] == Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-00000020;2'
[Sep 16 01:38:26] -- Executing [h@default:1] AGI("Local/55558600051@default-00000020;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Sep 16 01:38:26] -- <Local/55558600051@default-00000020;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Sep 16 01:38:27] == Manager 'sendcron' logged off from 127.0.0.1

I'm using an SIP account at who uses outbound proxy to make call.

Can you please help to solve this issue.

Registration String: register =>XXXXXXX:zzzzzzz@sipprovider/XXXXXXX

Account Entry:
[Voiptalk]
username=XXXXXXX
type=peer
trustrpid=no
sendrpid=yes
rtpkeepalive=5
secret=zzzzzzz
reinvite=yes
qualify=yes
insecure=very
host=dynamic
dtmfmode=rfc2833
disallow=all
canreinvite=no
fromuser=XXXXXXX
allow=ulaw
allow=alaw

Globals String: TESTSIPTRUNK= SIP/XXXXXXX

Dialplan Entry:
exten => _033XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _033XXXXXXXXX,2,Dial(${TESTSIPTRUNK}/${EXTEN:2},,To)
exten => _033XXXXXXXXX,3,Hangup

In the linux shell run this command:
asterisk -rx "sip show peers like Voiptalk" and post here the result.

Your dialplan does not match with the numbers you are dialing.
Number: 33235580657
Dialplan: 033XXXXXXXXX

You can remove the 0 from the dialplan, or add the 0 to the leads.