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Can not call

Added by Coman Tudor 4 months ago

I tried to figure out why i can't get into my campaign or just pass calls and i can't figure it out. (fresh install)
In the asterisk console is the only thing I can find, if someone has an idea would be great !

Regards
Tudor
GUI interface (no live call)
asterisk console


Replies (5)

RE: Can not call - Added by Wittie Manansala 4 months ago

Hi,

Please post your asterisk CLI from your agent login to making call/s for us to review your concern.

Thanks

RE: Can not call - Added by Coman Tudor 4 months ago

Hey, does this make the work:
i'll join the screen capture too.

vaglxc01*CLI> core set verbose 4
Console verbose was OFF and is now 4.
Manager 'sendcron' logged on from 127.0.0.1
Manager 'sendcron' logged off from 127.0.0.1
Manager 'updatecron' logged on from 127.0.0.1
Manager 'sendcron' logged on from 127.0.0.1
-- Called 99991869665930@default
-- Executing [99991869665930@default:1] Dial("Local/99991869665930@default-00000000;2", "SIP/1869665930@kamailio,,tTo") in new stack
Using SIP RTP CoS mark 5
-- Called SIP/1869665930@kamailio
Manager 'listencron' logged on from 127.0.0.1
-- SIP/kamailio-00000000 is ringing

Thanks
-- Local/99991869665930@default-00000000;1 is ringing
> 0x7f429400a230 -- Strict RTP learning after remote address set to: 192.168.1.200:30022
-- SIP/kamailio-00000000 answered Local/99991869665930@default-00000000;2
-- Local/99991869665930@default-00000000;1 answered
-- Executing [8600051@default:1] Konference("Local/99991869665930@default-00000000;1", "8600051,R") in new stack
Manager 'sendcron' logged off from 127.0.0.1
-- Channel SIP/kamailio-00000000 joined 'simple_bridge' basic-bridge <edf833ae-432a-4acf-9ca5-6a1faaab7f4a>
-- Channel Local/99991869665930@default-00000000;2 joined 'simple_bridge' basic-bridge <edf833ae-432a-4acf-9ca5-6a1faaab7f4a>
Manager 'sendcron' logged on from 127.0.0.1
Manager 'sendcron' logged off from 127.0.0.1
[Sep 27 09:21:08] WARNING[1872][C-00000001]: file.c:774 ast_openstream_full: File confbridge-join does not exist in any format
Manager 'sendcron' logged on from 127.0.0.1
-- Called 8600051@default
-- Executing [8600051@default:1] Konference("Local/8600051@default-00000001;2", "8600051,R") in new stack
-- Local/8600051@default-00000001;1 answered
-- Executing [33611942557@default:1] AGI in new stack
Manager 'sendcron' logged off from 127.0.0.1
-- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=12904133))
-- <Local/8600051@default-00000001;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing [33611942557@default:2] Dial("Local/8600051@default-00000001;1", "SIP/7@tudor,,tTo") in new stack
Using SIP RTP CoS mark 5
-- Called SIP/7@tudor
[Sep 27 09:21:23] WARNING[1872][C-00000001]: file.c:774 ast_openstream_full: File confbridge-join does not exist in any format
[Sep 27 09:21:23] WARNING[1919][C-00000002]: file.c:774 ast_openstream_full: File confbridge-join does not exist in any format
Manager 'sendcron' logged on from 127.0.0.1
-- Called 58600051@default
-- Executing [58600051@default:1] Konference("Local/58600051@default-00000002;2", "8600051,qLR") in new stack
-- Local/58600051@default-00000002;1 answered
-- Executing [8309@default:1] Answer("Local/58600051@default-00000002;1", "") in new stack
Manager 'sendcron' logged off from 127.0.0.1
-- Executing [8309@default:2] Monitor("Local/58600051@default-00000002;1", "wav,20220927-092123_611942557_12904133_tudor") in new stack
-- Executing [8309@default:3] Wait("Local/58600051@default-00000002;1", "3600") in new stack
[Sep 27 09:21:29] WARNING[1745]: chan_sip.c:4072 retrans_pkt: Retransmission timeout reached on transmission 46f1e60216c88b6d4e6f571d14f3e200@192.168.1.200:5070 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6401ms with no response
[Sep 27 09:21:29] WARNING[1745]: chan_sip.c:4096 retrans_pkt: Hanging up call 46f1e60216c88b6d4e6f571d14f3e200@192.168.1.200:5070 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
Everyone is busy/congested at this time (1:0/0/1)
-- Executing [33611942557@default:3] Hangup("Local/8600051@default-00000001;1", "") in new stack
Spawn extension (default, 33611942557, 3) exited non-zero on 'Local/8600051@default-00000001;1'
[Sep 27 09:21:29] WARNING1918[C-00000003]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
-- Executing [h@default:1] AGI") in new stack
-- <Local/8600051@default-00000001;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----18-----CHANUNAVAIL---------------) completed, returning 0
[Sep 27 09:21:29] WARNING1919[C-00000002]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
-- Executing [h@default:1] AGI") in new stack
-- <Local/8600051@default-00000001;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----18--------------------) completed, returning 0
[Sep 27 09:21:29] WARNING1872[C-00000001]: file.c:774 ast_openstream_full: File confbridge-leave does not exist in any format
[Sep 27 09:21:29] WARNING1926[C-00000004]: file.c:774 ast_openstream_full: File confbridge-leave does not exist in any format

RE: Can not call - Added by Jackie Alfonso 4 months ago

Hi,

We've noticed this on your CLI Called SIP/7@tudor and that is incorrect.

Please double check your dial plan and coordinate with your VoIP provider what dial format are they accepting.

Thank you!

RE: Can not call - Added by Coman Tudor 4 months ago

Jackie Alfonso wrote in RE: Can not call:

Hi,

We've noticed this on your CLI Called SIP/7@tudor and that is incorrect.

Please double check your dial plan and coordinate with your VoIP provider what dial format are they accepting.

Thank you!

Hi,

Tell me if I'm wrong, but when you create the carrer goautodial configurate everything alone, so in my dial when it sais SIP@tudor i understand that hes calling with the account entry, i may miss understood and i have to change this, idk.

Thanks !

RE: Can not call - Added by Jackie Alfonso 4 months ago

Coman Tudor wrote in RE: Can not call:

Jackie Alfonso wrote in RE: Can not call:

Hi,

We've noticed this on your CLI Called SIP/7@tudor and that is incorrect.

Please double check your dial plan and coordinate with your VoIP provider what dial format are they accepting.

Thank you!

Hi,

Tell me if I'm wrong, but when you create the carrer goautodial configurate everything alone, so in my dial when it sais SIP@tudor i understand that hes calling with the account entry, i may miss understood and i have to change this, idk.

Thanks !

Hi,

Base on your screenshot your dial plan seems correct. please try to make some test call again and provide the asterisk CLI logs if you encounter any issues again.

Thank you!

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