Dial Method:Manual works but others not

[cthax]

Hi i am a newbie in vicidial and a freelancer setup our trunk and dialplan

it works manual but the other versions not you can find below the report
but one quest more is why is there in the report a russion static ip and why it said wrong password? i dont understand why the ip is there

my system is:

Version: 2.14b0.5
SVN Version: 2832
DB Schema Version: 1520
DB Schema Update Date: 2017-10-05 13:07:14
Password Encryption: DISABLED - S1 - C1
Auto User-add Value: 101
Recording Prompt Count: 0
Install Date: 2017-10-05
Phone Codes: 1084 - 42588 - 0 - 0 - 0 - 0 - 0
Today System Stats: 8 - 0 - 3 - 11 - 3 - 1

Code: Select all

[Oct  9 10:41:43] NOTICE[2341]: chan_sip.c:28490 handle_request_register: Registration from '104 <sip:104@88.247.205.248>' failed for '46.29.162.7:35600' - Wrong password
[Oct  9 10:42:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct  9 10:42:01]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct  9 10:42:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct  9 10:42:01]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct  9 10:42:04] NOTICE[2341]: chan_sip.c:28490 handle_request_register: Registration from '103 <sip:103@88.247.205.248>' failed for '46.29.162.7:53973' - Wrong password
[Oct  9 10:42:06]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct  9 10:42:06]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct  9 10:42:11] NOTICE[2341]: chan_sip.c:28490 handle_request_register: Registration from '106 <sip:106@88.247.205.248>' failed for '46.29.162.7:45962' - Wrong password
[Oct  9 10:42:18] NOTICE[2341]: chan_sip.c:28490 handle_request_register: Registration from '101 <sip:101@88.247.205.248>' failed for '46.29.162.7:48605' - Wrong password


[dhanapathy]

hi,
someone trying to register sip account (trying to hack your asterisk), i guess your system is open to public network. please disable NAT, read and install fail2ban first.

[cthax]

hi,
someone trying to register sip account (trying to hack your asterisk), i guess your system is open to public network. please disable NAT, read and install fail2ban first.

thanks for advise i try it immediatly

[cthax]

know manual dial method isnt working too the report is below,

question:whi is the dialer calling 22 before (mentioned in red) the german number i want to call ? is that necessary?

in germany are numbers
49 (country code)
(x) (xx) (xxx) (xxxx) (area code)
(xxxx) (xxxxx) (xxxxxx) (xxxxxxx) (xxxxxxx) (xxxxxxxx) (xxxxxxxxx) (xxxxxxxxxx) (xxxxxxxxxxxxxxx) (rest of number)
i mean the number can be 5 digits lenght so can be 13 digits lenght.


all numbers that i want to call are

49xxxxxxx
49xxxxxxxx
49xxxxxxxxx
49xxxxxxxxxx
49xxxxxxxxxxx
49xxxxxxxxxxxx
49xxxxxxxxxxxxx

Code: Select all

Oct  9 15:21:38] Asterisk 11.25.1-vici, Copyright (C) 1999 - 2013 Digium, Inc. and others.
[Oct  9 15:21:38] Created by Mark Spencer <markster@digium.com>
[Oct  9 15:21:38] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
[Oct  9 15:21:38] This is free software, with components licensed under the GNU General Public
[Oct  9 15:21:38] License version 2 and other licenses; you are welcome to redistribute it under
[Oct  9 15:21:38] certain conditions. Type 'core show license' for details.
[Oct  9 15:21:38] =========================================================================
[Oct  9 15:21:38] Connected to Asterisk 11.25.1-vici currently running on vicibox (pid = 2116)
[Oct  9 15:21:43]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct  9 15:21:43]   == Using SIP RTP CoS mark 5
[Oct  9 15:21:46]        > Channel SIP/8001-00000011 was answered
[Oct  9 15:21:46]     -- Executing [8600051@default:1] MeetMe("SIP/8001-00000011", "8600051,F") in new stack
[Oct  9 15:21:46]   == Parsing '/etc/asterisk/meetme.conf': Found
[Oct  9 15:21:46]   == Parsing '/etc/asterisk/meetme-vicidial.conf': Found
[Oct  9 15:21:46]     -- Created MeetMe conference 1023 for conference '8600051'
[Oct  9 15:21:46]     -- <SIP/8001-00000011> Playing 'conf-onlyperson.gsm' (language 'en')
[Oct  9 15:21:46]        > 0x7fbbbc02a930 -- Probation passed - setting RTP source address to 192.168.1.51:4980
[Oct  9 15:21:47]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct  9 15:21:51]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct  9 15:21:51]     -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000014;2", "8600051,F") in new stack
[Oct  9 15:21:51]        > Channel Local/8600051@default-00000014;1 was answered
[Oct  9 15:21:51]     -- Executing [[color=#FF0000]224949514682[/color]@default:1] AGI("Local/8600051@default-00000014;1", "agi://127.0.0.1:4577/call_log") in new stack
[Oct  9 15:21:51]     -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=100))
[Oct  9 15:21:51]     -- <Local/8600051@default-00000014;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct  9 15:21:51]     -- Executing [[color=#FF0000]224949514682[/color]@default:2] Dial("Local/8600051@default-00000014;1", "sip/49514682@testcarrier,55,o") in new stack
[Oct  9 15:21:51]   == Using SIP RTP CoS mark 5
[Oct  9 15:21:51]     -- Called sip/49514682@testcarrier
[Oct  9 15:21:51]        > 0x7fbbc8014ae0 -- Probation passed - setting RTP source address to 77.72.168.32:25926
[Oct  9 15:21:51]        > 0x7fbbc8014ae0 -- Probation passed - setting RTP source address to 77.72.168.32:25926
[Oct  9 15:21:52]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct  9 15:21:53]     -- SIP/testcarrier-00000012 is making progress passing it to Local/8600051@default-00000014;1
[Oct  9 15:21:53]        > 0x7fbbc8014ae0 -- Probation passed - setting RTP source address to 77.72.168.32:25926
[Oct  9 15:22:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct  9 15:22:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct  9 15:22:01]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct  9 15:22:01]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct  9 15:22:06]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct  9 15:22:06]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct  9 15:22:16]     -- Got SIP response 480 "Temporarily not available" back from 77.72.174.129:5060
[Oct  9 15:22:16]     -- SIP/testcarrier-00000012 is circuit-busy
[Oct  9 15:22:16]   == Everyone is busy/congested at this time (1:0/1/0)
[Oct  9 15:22:16]     -- Executing [224949514682@default:3] Hangup("Local/8600051@default-00000014;1", "") in new stack
[Oct  9 15:22:16]   == Spawn extension (default, 224949514682, 3) exited non-zero on 'Local/8600051@default-00000014;1'
[Oct  9 15:22:16]     -- Executing [h@default:1] AGI("Local/8600051@default-00000014;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----19-----CONGESTION----------") in new stack
[Oct  9 15:22:16]     -- <Local/8600051@default-00000014;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----19-----CONGESTION---------- completed, returning 0
[Oct  9 15:22:16]   == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00000014;2'
[Oct  9 15:22:16]     -- Executing [h@default:1] AGI("Local/8600051@default-00000014;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Oct  9 15:22:16]     -- <Local/8600051@default-00000014;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Oct  9 15:22:24]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct  9 15:22:24]     -- Manager 'sendcron' from 127.0.0.1, hanging up channel: SIP/8001-00000011
[Oct  9 15:22:24]     -- Hungup 'DAHDI/pseudo-864920242'
[Oct  9 15:22:24]   == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/8001-00000011'
[Oct  9 15:22:24]     -- Executing [h@default:1] AGI("SIP/8001-00000011", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Oct  9 15:22:24]     -- <SIP/8001-00000011>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[Oct  9 15:22:24]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct  9 15:22:24]     -- Executing [55558600051@default:1] MeetMeAdmin("Local/55558600051@default-00000015;2", "8600051,K") in new stack
[Oct  9 15:22:24] WARNING[31477][C-00000087]: app_meetme.c:5053 admin_exec: Conference number '8600051' not found!
[Oct  9 15:22:24]     -- Executing [55558600051@default:2] Hangup("Local/55558600051@default-00000015;2", "") in new stack
[Oct  9 15:22:24]   == Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-00000015;2'
[Oct  9 15:22:24]     -- Executing [h@default:1] AGI("Local/55558600051@default-00000015;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Oct  9 15:22:24]     -- <Local/55558600051@default-00000015;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[Oct  9 15:22:25]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct  9 15:22:25]   == Manager 'sendcron' logged off from 127.0.0.1
vicibox*CLI>
Disconnected from Asterisk server
[Oct  9 15:22:26] Asterisk cleanly ending (0).
[Oct  9 15:22:26] Executing last minute cleanups
vicibox:~ #


[dhanapathy]

hi, may be 22 is added in campaign prefix, and it's removed before dialing. prefix is not a problem. check with your provider regarding call disconnection. for predictive dialing, i see you missed tT option in dial cmd. so please modify your dialplan.