DIAL ALERT: Call Rejected: CHANUNAVAIL Cause: 20 - Subscribe

[marouen2008]

hello everyone
I installed it goautodial-2.0-final.iso I configured my carrier and I did a load of my codec-codec_g729 ast14-gcc4-glibc-pentium4.so but the problem that by passing a manual call the agent screen an error message: DIAL ALERT: Call Rejected: CHANUNAVAIL Cause: 20 - Subscriber absent.

[Mar 30 05:55:38] -- Registered SIP 'cc101' at 192.168.24.148 port 59884
[Mar 30 05:55:38] -- Saved useragent "X-Lite release 1002tx stamp 29712" for peer cc101
[Mar 30 05:55:38] NOTICE[2767]: chan_sip.c:13408 handle_response_peerpoke: Peer 'cc101' is now Reachable. (1ms / 2000ms)
[Mar 30 05:56:01] == Parsing '/etc/asterisk/manager.conf': [Mar 30 05:56:01] Found
[Mar 30 05:56:01] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 30 05:56:01] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 30 05:56:01] == Parsing '/etc/asterisk/manager.conf': [Mar 30 05:56:01] Found
[Mar 30 05:56:01] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 30 05:56:06] == Parsing '/etc/asterisk/manager.conf': [Mar 30 05:56:06] Found
[Mar 30 05:56:06] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 30 05:56:06] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 30 05:56:07] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 30 05:56:10] == Parsing '/etc/asterisk/manager.conf': [Mar 30 05:56:10] Found
[Mar 30 05:56:10] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 30 05:56:10] > Channel SIP/cc101-00000000 was answered.
[Mar 30 05:56:10] -- Executing [8600051@default:1] MeetMe("SIP/cc101-00000000", "8600051|F") in new stack
[Mar 30 05:56:10] == Parsing '/etc/asterisk/meetme.conf': [Mar 30 05:56:10] Found
[Mar 30 05:56:10] == Parsing '/etc/asterisk/meetme-vicidial.conf': [Mar 30 05:56:10] Found
[Mar 30 05:56:10] -- Created MeetMe conference 1023 for conference '8600051'
[Mar 30 05:56:10] -- <SIP/cc101-00000000> Playing 'conf-onlyperson' (language 'en')
[Mar 30 05:56:14] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 30 05:56:15] == Parsing '/etc/asterisk/manager.conf': [Mar 30 05:56:15] Found
[Mar 30 05:56:15] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 30 05:56:15] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-55d8,2", "8600051|F") in new stack
[Mar 30 05:56:15] > Channel Local/8600051@default-55d8,1 was answered.
[Mar 30 05:56:15] -- Executing [10033146301118@default:1] AGI("Local/8600051@default-55d8,1", "agi://127.0.0.1:4577/call_log") in new stack
[Mar 30 05:56:15] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Mar 30 05:56:15] -- Executing [10033146301118@default:2] MixMonitor("Local/8600051@default-55d8,1", "10033146301118-1333101375.1.wav|bq") in new stack
[Mar 30 05:56:15] -- Executing [10033146301118@default:3] Dial("Local/8600051@default-55d8,1", "SIP/TW1/0033146301118||tTor") in new stack
[Mar 30 05:56:15] WARNING[3892]: app_dial.c:1296 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Mar 30 05:56:15] == Everyone is busy/congested at this time (1:0/0/1)
[Mar 30 05:56:15] -- Executing [10033146301118@default:4] Hangup("Local/8600051@default-55d8,1", "") in new stack
[Mar 30 05:56:15] == Spawn extension (default, 10033146301118, 4) exited non-zero on 'Local/8600051@default-55d8,1'
[Mar 30 05:56:15] -- Executing [h@default:1] DeadAGI("Local/8600051@default-55d8,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Mar 30 05:56:15] == Begin MixMonitor Recording Local/8600051@default-55d8,1
[Mar 30 05:56:15] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Mar 30 05:56:15] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-55d8,2'
[Mar 30 05:56:15] -- Executing [h@default:1] DeadAGI("Local/8600051@default-55d8,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Mar 30 05:56:15] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Mar 30 05:56:15] == End MixMonitor Recording Local/8600051@default-55d8,1
[Mar 30 05:56:16] == Parsing '/etc/asterisk/manager.conf': [Mar 30 05:56:16] Found
[Mar 30 05:56:16] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 30 05:56:16] -- Executing [58600051@default:1] MeetMe("Local/58600051@default-8182,2", "8600051|Fmq") in new stack
[Mar 30 05:56:16] > Channel Local/58600051@default-8182,1 was answered.
[Mar 30 05:56:16] -- Executing [8309@default:1] Answer("Local/58600051@default-8182,1", "") in new stack
[Mar 30 05:56:16] -- Executing [8309@default:2] Monitor("Local/58600051@default-8182,1", "wav|20120330-115615_146301118") in new stack
[Mar 30 05:56:16] -- Executing [8309@default:3] Wait("Local/58600051@default-8182,1", "3600") in new stack
[Mar 30 05:56:19] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 30 05:56:20] == Manager 'sendcron' logged off from 127.0.0.1


and when I call directly from my xlite

[Mar 30 06:01:43] -- Executing [10033149305844@default:1] AGI("SIP/cc101-00000002", "agi://127.0.0.1:4577/call_log") in new stack
[Mar 30 06:01:43] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Mar 30 06:01:43] -- Executing [10033149305844@default:2] MixMonitor("SIP/cc101-00000002", "10033149305844-1333101703.6.wav|bq") in new stack
[Mar 30 06:01:43] -- Executing [10033149305844@default:3] Dial("SIP/cc101-00000002", "SIP/TW1/0033149305844||tTor") in new stack
[Mar 30 06:01:43] == Begin MixMonitor Recording SIP/cc101-00000002
[Mar 30 06:01:43] WARNING[5079]: app_dial.c:1296 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Mar 30 06:01:43] == Everyone is busy/congested at this time (1:0/0/1)
[Mar 30 06:01:43] -- Executing [10033149305844@default:4] Hangup("SIP/cc101-00000002", "") in new stack
[Mar 30 06:01:43] == Spawn extension (default, 10033149305844, 4) exited non-zero on 'SIP/cc101-00000002'
[Mar 30 06:01:43] -- Executing [h@default:1] DeadAGI("SIP/cc101-00000002", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Mar 30 06:01:43] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Mar 30 06:01:43] == End MixMonitor Recording SIP/cc101-00000002
go*CLI>



there's my carrier:

Account Entry :

[TW1]
username=054xxxx785
type=friend
secret=7010xxxx79
host=xxxxxx
fromuser=054xxxx785
context=a2billing
allow=g729, alaw, ulaw, gsm ,ulaw,alaw,ilbc,gsm,g723.1,g726,g729a
trustrpid = yes
sendrpid = yes
canreinvite = no


Protocol : SIP

Globals String : TW1=SIP/TW1

Dialplan Entry: exten => _1X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1X.,2,MixMonitor(${EXTEN}-${UNIQUEID}.wav,bq)
exten => _1X.,3,Dial(SIP/TW1/${EXTEN:1},,tTor)
exten => _1X.,4,Hangup


and just for inforamtion I call French numbers
I put dial code 10033

[williamconley]

1) Welcome to the party!

2) when you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3) You did not show your dialplan entry for this carrier. I note that "mixmonitor" is not a standard entry.

4) You have not shown whether this host is accessible to the asterisk server. If it is "unreachable", asterisk will not attempt to dial it. sip show peers and sip show registry are useful commands. Try them out.