Hi,
I changed the entry as advise but now i am getting this message.
DIAL ALERT:
Call Rejected: CONGESTION
Cause: 1 - Unallocated (unassigned) number.
CLI output
vicibox*CLI> sip show registry
Host Username Refresh State Reg. Time
85.232.50.152:5060 18826633333 105 Registered Fri, 17 Dec 2010 03:23:20
vicibox*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
gs102/gs102 (Unspecified) D N 0 UNKNOWN
201/201 192.168.1.100 D N 12972 OK (102 ms)
viva/18826633333 85.232.50.152 N 5060 OK (174 ms)
3 sip peers [Monitored: 2 online, 1 offline Unmonitored: 0 online, 0 offline]
[Dec 17 03:24:32] == Parsing '/etc/asterisk/manager.conf': [Dec 17 03:24:32] F ound
[Dec 17 03:24:32] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 17 03:24:35] > Channel SIP/201-00000006 was answered.
[Dec 17 03:24:35] -- Executing [8600051@default:1] MeetMe("SIP/201-00000006" , "8600051|F") in new stack
[Dec 17 03:24:35] == Parsing '/etc/asterisk/meetme.conf': [Dec 17 03:24:35] Fo und
[Dec 17 03:24:35] == Parsing '/etc/asterisk/meetme-vicidial.conf': [Dec 17 03: 24:35] Found
[Dec 17 03:24:35] -- Created MeetMe conference 1023 for conference '8600051'
[Dec 17 03:24:35] -- <SIP/201-00000006> Playing 'conf-onlyperson' (language 'en')
[Dec 17 03:24:37] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 17 03:24:41] == Parsing '/etc/asterisk/manager.conf': [Dec 17 03:24:41] F ound
[Dec 17 03:24:41] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 17 03:24:41] -- Executing [8600051@default:1] MeetMe("Local/8600051@def ault-102b,2", "8600051|F") in new stack
[Dec 17 03:24:41] > Channel Local/8600051@default-102b,1 was answered.
[Dec 17 03:24:41] -- Executing [914032830919@default:1] AGI("Local/8600051@d efault-102b,1", "agi://127.0.0.1:4577/call_log") in new stack
[Dec 17 03:24:41] -- AGI Script
agi://127.0.0.1:4577/call_log completed, ret urning 0
[Dec 17 03:24:41] -- Executing [914032830919@default:2] Dial("Local/8600051@ default-102b,1", "SIP/viva/14032830919||tTor") in new stack
[Dec 17 03:24:41] -- Called viva/14032830919
[Dec 17 03:24:41] -- SIP/viva-00000007 is circuit-busy
[Dec 17 03:24:41] == Everyone is busy/congested at this time (1:0/1/0)
[Dec 17 03:24:41] -- Executing [914032830919@default:3] Hangup("Local/860005 1@default-102b,1", "") in new stack
[Dec 17 03:24:41] == Spawn extension (default, 914032830919, 3) exited non-zer o on 'Local/8600051@default-102b,1'
[Dec 17 03:24:41] -- Executing [h@default:1] DeadAGI("Local/8600051@default- 102b,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1-----CON GESTION----------") in new stack
[Dec 17 03:24:42] -- AGI Script
agi://127.0.0.1:4577/call_log--HVcauses--PRI -----NODEBUG-----1-----CONGESTION---------- completed, returning 0
[Dec 17 03:24:42] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-102b,2'
[Dec 17 03:24:42] -- Executing [h@default:1] DeadAGI("Local/8600051@default- 102b,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-------- -------") in new stack
[Dec 17 03:24:42] -- AGI Script
agi://127.0.0.1:4577/call_log--HVcauses--PRI -----NODEBUG-----0--------------- completed, returning 0
[Dec 17 03:24:43] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 17 03:25:02] == Parsing '/etc/asterisk/manager.conf': [Dec 17 03:25:02] Found
[Dec 17 03:25:02] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 17 03:25:02] == Parsing '/etc/asterisk/manager.conf': [Dec 17 03:25:02] Found
[Dec 17 03:25:02] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 17 03:25:02] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 17 03:25:02] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 17 03:25:07] == Parsing '/etc/asterisk/manager.conf': [Dec 17 03:25:07] Found
[Dec 17 03:25:07] == Manager 'sendcron' logged on from 127.0.0.1
[Dec 17 03:25:07] == Manager 'sendcron' logged off from 127.0.0.1
CLI sip debug
<------------->
[Dec 17 13:03:13] --- (7 headers 0 lines) ---
[Dec 17 13:03:13]
<--- SIP read from 85.232.50.152:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK75094b17;rport
From: "M2171303120000060027" <sip:0000000000@192.168.1.101>;tag=as46041915
To: <sip:14033741732@85.232.50.152;cpd=on>
Contact: <sip:0000000000@192.168.1.101>
Call-ID:
547892702126711c74028558609f2402@192.168.1.101 CSeq: 102 INVITE
User-agent: Asterisk PBX
Max-Forwards: 69
Remote-Party-ID: "M2171303120000060027" <sip:0000000000@192.168.1.101>;privacy=off;screen=no
Date: Fri, 17 Dec 2010 07:33:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 237
v=0
o=root 2443 2443 IN IP4 192.168.1.101
s=session
c=IN IP4 192.168.1.101
t=0 0
m=audio 14686 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
[Dec 17 13:03:13] --- (15 headers 12 lines) ---
[Dec 17 13:03:13] Transmitting (NAT) to 85.232.50.152:5060:
ACK sip:14033741732@85.232.50.152;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK75094b17;rport
From: "M2171303120000060027" <sip:0000000000@192.168.1.101>;tag=as46041915
To: <sip:14033741732@85.232.50.152;cpd=on>
Contact: <sip:0000000000@192.168.1.101>
Call-ID:
547892702126711c74028558609f2402@192.168.1.101 CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "M2171303120000060027" <sip:0000000000@192.168.1.101>;privacy=off;screen=no
Content-Length: 0
---
[Dec 17 13:03:13] -- SIP/viva-00000001 is circuit-busy
[Dec 17 13:03:13] == Everyone is busy/congested at this time (1:0/1/0)
[Dec 17 13:03:13] -- Executing [914033741732@default:3] Hangup("Local/8600051@default-f324,1", "") in new stack
[Dec 17 13:03:13] == Spawn extension (default, 914033741732, 3) exited non-zero on 'Local/8600051@default-f324,1'
[Dec 17 13:03:13] -- Executing [h@default:1] DeadAGI("Local/8600051@default-f324,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1-----CONGESTION----------") in new stack
[Dec 17 13:03:13] Really destroying SIP dialog
'547892702126711c74028558609f2402@192.168.1.101' Method: INVITE
[Dec 17 13:03:13] -- AGI Script
agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Dec 17 13:03:13] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-f324,2'
[Dec 17 13:03:13] -- Executing [h@default:1] DeadAGI("Local/8600051@default-f324,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Dec 17 13:03:13] -- AGI Script
agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Dec 17 13:03:15] == Manager 'sendcron' logged off from 127.0.0.1
[Dec 17 13:03:28]
<--- SIP read from 192.168.1.100:32468 --->
<------------->
[Dec 17 13:03:33] Reliably Transmitting (NAT) to 85.232.50.152:5060:
OPTIONS sip:85.232.50.152;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK795134b4;rport
From: "asterisk" <sip:asterisk@192.168.1.101>;tag=as10254fd0
To: <sip:85.232.50.152;cpd=on>
Contact: <sip:asterisk@192.168.1.101>
Call-ID:
354cb85c1dc0eb4c2e9c8421736bb4be@192.168.1.101 CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 17 Dec 2010 07:33:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
---
[Dec 17 13:03:33]
<--- SIP read from 85.232.50.152:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK795134b4;rport
From: "asterisk" <sip:asterisk@192.168.1.101>;tag=as10254fd0
To: <sip:85.232.50.152;cpd=on>
Contact: <sip:asterisk@192.168.1.101>
Call-ID:
354cb85c1dc0eb4c2e9c8421736bb4be@192.168.1.101 CSeq: 102 OPTIONS
User-agent: Asterisk PBX
Max-Forwards: 69
Date: Fri, 17 Dec 2010 07:33:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
<------------->
[Dec 17 13:03:33] --- (13 headers 0 lines) ---
[Dec 17 13:03:33] Really destroying SIP dialog
'354cb85c1dc0eb4c2e9c8421736bb4be@192.168.1.101' Method: OPTIONS
[Dec 17 13:03:44] NOTICE[2509]: chan_sip.c:8028 sip_reregister: -- Re-registration for
18826633333@85.232.50.152 [Dec 17 13:03:44] REGISTER 13 headers, 0 lines
[Dec 17 13:03:44] Reliably Transmitting (NAT) to 85.232.50.152:5060:
REGISTER sip:85.232.50.152 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK76c81d26;rport
From: <sip:18826633333@85.232.50.152>;tag=as56fb2c83
To: <sip:18826633333@85.232.50.152>
Call-ID:
792e70fa79e9243c122e34e2236b477a@127.0.0.2 CSeq: 590 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="18826633333", realm="Sysmaster", algorithm=MD5, uri="sip:85.232.50.152", nonce="96a6c8431b643f39759f1fb3b855729b", response="25cebf4fc82aee05bb5abff2dded513e", opaque="bdb1e436290d395987f61987298d6475"
Expires: 120
Contact: <sip:s@192.168.1.101>
Event: registration
Content-Length: 0
---
[Dec 17 13:03:44]
<--- SIP read from 85.232.50.152:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK76c81d26;rport
From: <sip:18826633333@85.232.50.152>;tag=as56fb2c83
To: <sip:18826633333@85.232.50.152>
Call-ID:
792e70fa79e9243c122e34e2236b477a@127.0.0.2 CSeq: 590 REGISTER
User-agent: Asterisk PBX
Max-Forwards: 69
Authorization: Digest username="18826633333", realm="Sysmaster", algorithm="MD5", uri="sip:85.232.50.152", nonce="96a6c8431b643f39759f1fb3b855729b", response="25cebf4fc82aee05bb5abff2dded513e", opaque="bdb1e436290d395987f61987298d6475"
Expires: 120
Contact: <sip:s@192.168.1.101>
Event: registration
Content-Length: 0
Proxy-Authenticate: Digest realm="Sysmaster", nonce="49595794fce73ca55ec3f08da83c029e", opaque="752946ef9d6a1ac536bcad1d91ce6b85", uri="sip:85.232.50.152"
<------------->
[Dec 17 13:03:44] --- (14 headers 0 lines) ---
[Dec 17 13:03:44] Responding to challenge, registration to domain/host name 85.232.50.152
[Dec 17 13:03:44] REGISTER 13 headers, 0 lines
[Dec 17 13:03:44] Reliably Transmitting (NAT) to 85.232.50.152:5060:
REGISTER sip:85.232.50.152 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK6877d07b;rport
From: <sip:18826633333@85.232.50.152>;tag=as5f0a9bdd
To: <sip:18826633333@85.232.50.152>
Call-ID:
792e70fa79e9243c122e34e2236b477a@127.0.0.2 CSeq: 591 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="18826633333", realm="Sysmaster", algorithm=MD5, uri="sip:85.232.50.152", nonce="49595794fce73ca55ec3f08da83c029e", response="d84cef028afea486b9bce8ae1397a57a", opaque="752946ef9d6a1ac536bcad1d91ce6b85"
Expires: 120
Contact: <sip:s@192.168.1.101>
Event: registration
Content-Length: 0
---
[Dec 17 13:03:44]
<--- SIP read from 85.232.50.152:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK6877d07b;rport
From: <sip:18826633333@85.232.50.152>;tag=as5f0a9bdd
To: <sip:18826633333@85.232.50.152>
Call-ID:
792e70fa79e9243c122e34e2236b477a@127.0.0.2 CSeq: 591 REGISTER
User-agent: Asterisk PBX
Max-Forwards: 69
Expires: 120
Contact: <sip:s@192.168.1.101>
Event: registration
Content-Length: 0
<------------->
[Dec 17 13:03:44] --- (12 headers 0 lines) ---
[Dec 17 13:03:44] Scheduling destruction of SIP dialog
'792e70fa79e9243c122e34e2236b477a@127.0.0.2' in 32000 ms (Method: REGISTER)
[Dec 17 13:03:44] NOTICE[2509]: chan_sip.c:13365 handle_response_register: Outbound Registration: Expiry for 85.232.50.152 is 120 sec (Scheduling reregistration in 105 s)
vicibox*CLI> sip debug off
Usage: sip set debug
Enables dumping of SIP packets for debugging purposes
sip set debug ip <host[:PORT]>
Enables dumping of SIP packets to and from host.
sip set debug peer <peername>
Enables dumping of SIP packets to and from host.
Require peer to be registered.
The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead.
vicibox*CLI> quit[/quote]
Please help me in coming out of it, i am really getting worried, what to do
