Call Rejected: CONGESTION Cause: 16 - Normal call clearing.

[x44]

I am getting this message while dialing out. The server would not dial out.

This setup used to work perfectly, but broke down 3 days ago. Not sure it it is my fault or not

I reinstalled asterisk and MySQL database, and now the call connects, but there is no audio [separate issue] and after 3 seconds hangs up, giving the same message.

All 9 screens are running

Since it worked before, I am eager to reinstall appropriate module responsible for this mess, but I am not sure which one it is.

This is a test box, so I am free to do with is whatever is needed.

Here is my setup, output of a call and a file with error message

[MainCarrier]
trustrpid=yes
generaterpid=yes
sendrpid=yes
type=friend
host=xxx.xxx.xx.xx
username=xxxxxxxxxxxxxx
secret=xxxxxxxxxxx
fromuser=xxxxxxxxxxxxxx
disallow=all
allow=ulaw
allow=alaw
dtmfmode=rfc2833
qualify=yes
externip=xx.xx.xxx.xxx
nat=yes
canreinvite=no


DIALPLAN
exten => _97771NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _97771NXXNXXXXXX,2,Dial(SIP/${EXTEN:4}@MainCarrier,,Tor)
exten => _97771NXXNXXXXXX,3,Hangup



-- Created MeetMe conference 1023 for conference '8600050'
[Apr 2 22:54:29] -- <SIP/212-00000011> Playing 'conf-onlyperson' (language 'en')
[Apr 2 22:54:31] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 2 22:54:33] ERROR[5148]: chan_iax2.c:4354 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 123.238.229.65 in the calltokenoptional list or setting user cc301 requirecalltoken=no
[Apr 2 22:54:37] == Parsing '/etc/asterisk/manager.conf': [Apr 2 22:54:37] Found
[Apr 2 22:54:37] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 2 22:54:37] -- Executing [8600050@default:1] MeetMe("Local/8600050@default-6c54,2", "8600050|F") in new stack
[Apr 2 22:54:37] > Channel Local/8600050@default-6c54,1 was answered.
[Apr 2 22:54:37] -- Executing [1xxxxxxxxx@default:1] AGI("Local/8600050@default-6c54,1", "agi://127.0.0.1:4577/call_log") in new stack
[Apr 2 22:54:37] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Apr 2 22:54:37] -- Executing [1xxxxxxxxx@default:2] GotoIf("Local/8600050@default-6c54,1", "0?NODIAL") in new stack
[Apr 2 22:54:37] -- Executing [1xxxxxxxxx@default:3] GotoIf("Local/8600050@default-6c54,1", "0?NODIAL") in new stack
[Apr 2 22:54:37] -- Executing [1xxxxxxxxx@default:4] GotoIf("Local/8600050@default-6c54,1", "0?NODIAL") in new stack
[Apr 2 22:54:37] -- Executing [1xxxxxxxxx@default:5] GotoIf("Local/8600050@default-6c54,1", "0?NODIAL") in new stack
[Apr 2 22:54:37] -- Executing [1xxxxxxxxx@default:6] Dial("Local/8600050@default-6c54,1", "SIP/19175686588@MainCarrier||Tor") in new stack
[Apr 2 22:54:37] -- Called 19175686588@MainCarrier
[Apr 2 22:54:39] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 2 22:54:49] NOTICE[5176]: chan_sip.c:3245 auto_congest: Auto-congesting SIP/MainCarrier-00000012
[Apr 2 22:54:49] -- SIP/MainCarrier-00000012 is circuit-busy
[Apr 2 22:54:49] == Everyone is busy/congested at this time (1:0/1/0)
[Apr 2 22:54:49] -- Executing [1xxxxxxxxx@default:7] Hangup("Local/8600050@default-6c54,1", "") in new stack
[Apr 2 22:54:49] == Spawn extension (default, 1xxxxxxxxx, 7) exited non-zero on 'Local/8600050@default-6c54,1'
[Apr 2 22:54:49] -- Executing [h@default:1] DeadAGI("Local/8600050@default-6c54,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CONGESTION----------") in new stack
[Apr 2 22:54:49] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 2 22:54:49] == Spawn extension (default, 8600050, 1) exited non-zero on 'Local/8600050@default-6c54,2'
[Apr 2 22:54:49] -- Executing [h@default:1] DeadAGI("Local/8600050@default-6c54,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr 2 22:54:49] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0



root@ubuntu:~# locate chan_sip.c
/usr/src/asterisk/asterisk-1.4.21.2-vici/channels/chan_sip.c
/usr/src/asterisk/asterisk-1.4.27.1/asterisk-1.4.27.1-vici/channels/chan_sip.c
/usr/src/asterisk/asterisk-1.4.27.1/asterisk-1.4.27.1-vici/channels/chan_sip.c.orig
/usr/src/asterisk/asterisk-1.4.27.1/channels/chan_sip.c
/usr/src/asterisk/asterisk-1.4.27.1/channels/chan_sip.c.orig


---------------------------------------------------------------------------------

UPDATE 3 DAYS LATER

Still no replies. Nobody even asked me to post installation method, build#, hardware vendor, and monitor resolution.

I solved the problem by temporarily by setting up a IAX carrier with IAX phone. Calls connect with clear sound both ways.

SIP phone AND carrier do not work - call rings and connects, but no sound both ways, and the same error is flashing after 30 sec.


I will be hacking away at SIP carrier setup
AMD-tk-55, 1.5 GB RAM, 45GB HDD
PUBLIC IP
scratch setup
http://wiki.vicidial.org/index.php/VICI:UbuntuInstall

[williamconley]

Auto-congesting

post results from "sip show peers"

if your carrier is "unreachable", change qualify=no

but this is ordinarily an indication that the network has a problem. packets are dropping or very slow to return.