CONGECTION status

[achillf]

All dialer calls in carrier goes to CONGECTION and when i try to call manual i get this message: Call Rejected: CONGESTION Cause: 21 - Call rejected.

In sip show peers carrier is status OK, my server has internet connection (i ping google for example), there is no firewall activated, gateway is from the same dhcp table.

VERSION: 2.6-380a
BUILD: 121029-0109
© 2012 ViciDial Group

[ambiorixg12]

Call Rejected: CONGESTION Cause: 21 - Call rejected.

This require you show you Asterisk CLI at the moment of the call, I dont know if that status showed on the vicidial GUI match the Asterisk ${DIALSTATUS} variable and ${HANGUPCAUSE} variable but if this is the case

CONGESTION: Congestion. This status is usually a sign that the dialled number is not recognized.

Hangup cause 21 : Is 21. Call Rejected, you carrier reject the call most of the time this could caused due to authentication issue, you re not allowed to dial to that number

Bu again Asterisk CLI is needed

[achillf]

on login in campaign:

[Dec 27 20:43:09] > 0x21cc380 -- Strict RTP learning after remote address set to: XX.XXX.X.XXX:8000
[Dec 27 20:43:09] -- SIP/100-0000000c answered
[Dec 27 20:43:09] -- Executing [8600051@default:1] MeetMe("SIP/100-0000000c", "8600051,F") in new stack
[Dec 27 20:43:09] -- Created MeetMe conference 1023 for conference '8600051'
[Dec 27 20:43:09] -- <SIP/100-0000000c> Playing 'conf-onlyperson.gsm' (language 'en')
[Dec 27 20:43:09] > 0x21cc380 -- Strict RTP switching to RTP target address XX.XXX.X.XXX:8000 as source
[Dec 27 20:43:10] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 27 20:43:14] > 0x21cc380 -- Strict RTP learning complete - Locking on source address XX.XXX.X.XXX:8000
[Dec 27 20:44:02] == Manager 'sendcron' logged on from 127.0.0.1

When i call manual:

SIP/100-0000000c answered
[Dec 27 20:43:09] -- Executing [8600051@default:1] MeetMe("SIP/100-0000000c", "8600051,F") in new stack
[Dec 27 20:43:09] -- Created MeetMe conference 1023 for conference '8600051'
[Dec 27 20:43:09] -- <SIP/100-0000000c> Playing 'conf-onlyperson.gsm' (language 'en')
[Dec 27 20:43:09] > 0x21cc380 -- Strict RTP switching to RTP target address XX.XXX.X.XXX:8000 as source
[Dec 27 20:43:10] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 27 20:43:14] > 0x21cc380 -- Strict RTP learning complete - Locking on source address XX.XXX.X.XXX:8000
[Dec 27 20:44:02] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 27 20:44:02] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 27 20:44:07] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 27 20:44:07] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 27 20:44:20] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 27 20:44:20] -- Called 8600051@default
[Dec 27 20:44:20] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-0000000a;2", "8600051,F") in new stack
[Dec 27 20:44:20] -- Local/8600051@default-0000000a;1 answered
[Dec 27 20:44:20] -- Executing [9230XXXXXXXXXX@default:1] AGI("Local/8600051@default-0000000a;1", "agi://127.0.0.1:4577/call_log") in new stack
[Dec 27 20:44:20] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=SAMACAMP))
[Dec 27 20:44:20] -- <Local/8600051@default-0000000a;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Dec 27 20:44:20] -- Executing [9230XXXXXXXXXX@default:2] Dial("Local/8600051@default-0000000a;1", "SIP/vlg_24/30XXXXXXXXXX,90,tTor") in new stack
[Dec 27 20:44:20] == Using SIP RTP CoS mark 5
[Dec 27 20:44:20] -- Called SIP/vlg_24/30XXXXXXXXXX
[Dec 27 20:44:20] NOTICE[1880][C-00000017]: chan_sip.c:24042 handle_response_invite: Failed to authenticate on INVITE to '"M2272044190000000008" <sip:30XXXXXXXXXX@XXX.XXX.XXX.X>;tag=as3569eefd'
[Dec 27 20:44:20] -- SIP/vlg_24-0000000d is circuit-busy
[Dec 27 20:44:20] == Everyone is busy/congested at this time (1:0/1/0)
[Dec 27 20:44:20] -- Executing [9230XXXXXXXXXX@default:3] Hangup("Local/8600051@default-0000000a;1", "") in new stack
[Dec 27 20:44:20] == Spawn extension (default, 9230XXXXXXXXXX, 3) exited non-zero on 'Local/8600051@default-0000000a;1'
[Dec 27 20:44:20] -- Executing [h@default:1] AGI("Local/8600051@default-0000000a;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----CONGESTION---------------SIP 401 Unauthorized)") in new stack
[Dec 27 20:44:20] -- <Local/8600051@default-0000000a;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -------SIP 401 Unauthorized) completed, returning 0
[Dec 27 20:44:20] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-0000000a;2'
[Dec 27 20:44:20] WARNING[18264][C-00000016]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Dec 27 20:44:20] -- Executing [h@default:1] AGI("Local/8600051@default-0000000a;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21--------------------)") in new stack
[Dec 27 20:44:20] -- <Local/8600051@default-0000000a;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------) completed, returning 0

[ambiorixg12]

As I said on my previous post hangup cause 21, is call rejected. Carrier vlg_24 is rejecting your call and the logs show it clearly

[Dec 27 20:44:20] NOTICE[1880][C-00000017]: chan_sip.c:24042 handle_response_invite: Failed to authenticate on INVITE to '"M2272044190000000008" <sip:30XXXXXXXXXX@XXX.XXX.XXX.X>;tag=as3569eefd'

So make sure you are sending the correct authentication information on your INVITE, or ask them what are you missing

[williamconley]

Wrong user/pass (or field names, or attempting from an IP that's not allowed to make calls) OR if you include user/pass information when you are authenticating via IP address, you can cause authentication failure.

Check the carrier's documentation for the proper method for authentication. Some have dial prefixes, sip headers, all sorts of methods are possible: There are technically no rules for this, but there are some "norms".