calls dialed with no agents online.
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I appreciate any insight offered as to why would vicidial call leads with no agent active. There are no records in the logs of any agent being active and no call logs viewable from admin panel. We were doing some test to login using zoiper but the conference call wasn't accepted. The carrier call logs show the thousands of calls made. also the asterisk log showed the calls.
Dialplan entry:
I rotated the log since it was over 9gb . I have a few errors.
thanks in advance for any help
Dialplan entry:
- Code: Select all
exten => _9X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9X.,2,Dial(${Telnyx}/${EXTEN:1},60,tTor)
exten => _9X.,3,Hangup
I rotated the log since it was over 9gb . I have a few errors.
- Code: Select all
[Feb 13 14:22:29] NOTICE[14674][C-000dcf24] chan_sip.c: Failed to authenticate device <sip:1002@[myexternalip]>;tag=1610631914 for INVITE, code = -1
[Feb 13 14:22:59] VERBOSE[14674][C-000dcf25] netsock2.c: Using SIP RTP CoS mark 5
[Feb 13 14:22:59] NOTICE[14674][C-000dcf25] chan_sip.c: No compatible codecs, not accepting this offer!
[Feb 13 14:22:59] VERBOSE[14674][C-000dcf25] netsock2.c: Using SIP RTP CoS mark 5
[Feb 13 14:22:59] NOTICE[14674][C-000dcf25] chan_sip.c: Call from '1001' (185.224.128.188:5064) to extension '82011421233015104' rejected because extension not found in context 'default'.
[Feb 13 14:23:01] WARNING[14674] chan_sip.c: Retransmission timeout reached on transmission 1469080746-302994384-1234257338 for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
thanks in advance for any help