Always receives 486 congestion after ringing

I am tackled by a strange problem. There is one and only one extension that cannot be dialed, there was no problem in the past.

[color=blue] – Executing Dial(“SIP/8888010139-08c3c2b0”, “SIP/8888010005”) in new stack
– Called 8888010005channel
– SIP/8888010005-08c1d890 is ringing
– Got SIP response 486 “Busy Here” back from xxx.xxx.xxx.xxx
– SIP/8888010005-08c1d890 is busy
== Everyone is busy/congested at this time (1:1/0/0)
– Executing Macro(“SIP/8888010139-08c3c2b0”, “outisbusy”) in new stack
– Executing Playback(“SIP/8888010139-08c3c2b0”, “all-circuits-busy-now”) in new stack
– Playing ‘all-circuits-busy-now’ (language ‘en’)
== Spawn extension (macro-outisbusy, s, 1) exited non-zero on ‘SIP/8888010139-08c3c2b0’ in macro ‘outisbusy’
== Spawn extension (macro-outisbusy, s, 1) exited non-zero on ‘SIP/8888010139-08c3c2b0’[/color]

It is very strange that 486 congestion coming up after ringing. Then:

[color=blue]asterisk1*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message
192.168.123.173 (None) 1389776994@ 00101/00782 unkn No Rx: REGISTER[/color]

Ooops! It is always there. This must be a part of the problem!

[color=blue]asterisk1CLI> sip show channel 1389776994@192.168.123.173
asterisk1
CLI>

  • SIP CallI>
    Direction: Incoming
    Call-ID: 1389776994@192.168.123.173
    Our Codec Capability: 1295
    Non-Codec Capability: 1
    Their Codec Capability: 0
    Joint Codec Capability: 0
    Format unknown
    Theoretical Address: 192.168.123.173:5060
    Received Address: xxx.xxx.xxx.xxx:35027 ; due to an encryption application, nothing wrong to other encrypted extensions
    NAT Support: Always
    Audio IP: xxx.xxx.xxx.xxx (local)
    Our Tag: as029c2adb
    Their Tag: 1295721275
    SIP User agent: GCE-4019N-L135 2007/01/27 09:27:53
    Need Destroy: 0
    Last Message: Rx: REGISTER
    Promiscuous Redir: No
    Route: N/A
    DTMF Mode: rfc2833
    SIP Options: (none)[/color]

Then sip show history:

[color=blue]331. Rx REGISTER / 803 REGISTER /sip:GCEREAL2
332. TxResp SIP/2.0 / 803 REGISTER
333. SchedDestroy 15000 ms
334. Rx REGISTER / 803 REGISTER /sip:GCEREAL2
335. TxResp SIP/2.0 / 803 REGISTER
336. SchedDestroy 15000 ms
337. Rx REGISTER / 803 REGISTER /sip:GCEREAL2
338. TxResp SIP/2.0 / 803 REGISTER
339. SchedDestroy 15000 ms
340. Rx REGISTER / 803 REGISTER /sip:GCEREAL2
341. TxResp SIP/2.0 / 803 REGISTER
342. SchedDestroy 15000 ms
343. Rx REGISTER / 803 REGISTER /sip:GCEREAL2
344. TxResp SIP/2.0 / 803 REGISTER
345. SchedDestroy 15000 ms
346. Rx REGISTER / 804 REGISTER /sip:GCEREAL2
347. TxResp SIP/2.0 / 804 REGISTER
348. SchedDestroy 15000 ms
349. Rx REGISTER / 804 REGISTER /sip:GCEREAL2
350. TxResp SIP/2.0 / 804 REGISTER
351. SchedDestroy 15000 ms
352. Rx REGISTER / 804 REGISTER /sip:GCEREAL2
353. TxResp SIP/2.0 / 804 REGISTER
354. SchedDestroy 15000 ms
355. Rx REGISTER / 804 REGISTER /sip:GCEREAL2
356. TxResp SIP/2.0 / 804 REGISTER
357. SchedDestroy 15000 ms
358. Rx REGISTER / 804 REGISTER /sip:GCEREAL2
359. TxResp SIP/2.0 / 804 REGISTER
360. SchedDestroy 15000 ms
361. Rx REGISTER / 804 REGISTER /sip:GCEREAL2
362. TxResp SIP/2.0 / 804 REGISTER
363. SchedDestroy 15000 ms
364. Rx REGISTER / 804 REGISTER /sip:GCEREAL2
365. TxResp SIP/2.0 / 804 REGISTER
366. SchedDestroy 15000 ms[/color]

sip debug peer 8888010005:


[color=blue]Scheduling destruction of call ‘1389776994@192.168.123.173’ in 15000 ms
asterisk1*CLI>
<-- SIP read from xxx.xxx.xxx.xxx:35027:
REGISTER sip:GCEREAL2 SIP/2.0
Via: SIP/2.0/UDP 192.168.123.173:5060;branch=z9hG4bK848815562;rport
From: sip:GCEUSER2@GCEREAL2;tag=1295721275
To: sip:GCEUSER2@GCEREAL2
Call-ID: 1389776994@192.168.123.173
CSeq: 810 REGISTER
Contact: sip:GCEUSER2@192.168.123.173:5060
Supported: 100rel,replaces
Max-Forwards: 70
User-Agent: GCE-4019N-L135 2007/01/27 09:27:53
Expires: 3600
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE
Content-Length: 0

— (13 headers 0 lines)—
Using latest REGISTER request as basis request
Sending to 192.168.123.173 : 5060 (NAT)
Transmitting (NAT) to xxx.xxx.xxx.xxx:35027:
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 192.168.123.173:5060;branch=z9hG4bK848815562;received=xxx.xxx.xxx.xxx;rport=35027
From: sip:GCEUSER2@GCEREAL2;tag=1295721275
To: sip:GCEUSER2@GCEREAL2;tag=as029c2adb
Call-ID: 1389776994@192.168.123.173
CSeq: 810 REGISTER
User-Agent: zita 0.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:GCEUSER2@xxx.xxx.xxx.xxx
Content-Length: 0[/color]

Actually this extension has registered on Asterisk.

I tried to restart amportal and even reboot the box, but the problem still exists.

Any one tell me why this extension keeps on sending registered request, blocking all incoming invite after ringing and how can I stop this sip channel?