QUESTION why PJSIP registered with port 10XX?

windpoint

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I installed Incredible PBX 13 on HiFormance
So it come with public IP. sip phone(cisco 7960) is behind NAT
when release "pjsip show endpoints" in asterisk, I see this

Endpoint: 1008/1008 Not in use 0 of inf
InAuth: 1008-auth/1008
Aor: 1008 1
Contact: 1008/sip:[email protected]:1033 b8efecd32b Avail 148.402


I believe I set the port to 5061 in Setting/PJSIP

This device is a sip version of cisco 7960. It can now making calls with no known issue. But it wont ring wen I call from other working extension or from IVR.
The phone got automatically disconnected with message "the person at extension at 1008 is not available"

The phone works for sip extensions.
Extensions 1008 works on soft sip phone

Any way I can bind the port to 5061?
 
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windpoint

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Below is from the logfile
1002 is yealink Sip phone, no issue with calling
1008 is Cisco 7960 with sip. no issue with making calls, but got no ring,

both extensions are on PJSIP

XXX.XXX.XXX.XXX is the PBX IP, public ip
YYY.YYY.YYY.YYY Phone wan IP, Phones are behind NAT with IP 10.10.10.xxx


Any help are highly appreciated.


[2018-07-26 14:52:08] VERBOSE[14848] res_pjsip_logger.c: <--- Received SIP request (651 bytes) from UDP:YYY.YYY.YYY.YYY:5061 --->
BYE sip:XXX.XXX.XXX.XXX:5061 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.165:5061;branch=z9hG4bK3718025555
From: "Peter J" <sip:[email protected]>;tag=3161788746
To: <sip:[email protected]>;tag=4e31566c-4dcf-44fa-834d-03bb16cee9e0
Call-ID: [email protected]
CSeq: 3 BYE
Contact: <sip:[email protected]:5061>
Authorization: Digest username="1002", realm="asterisk", nonce="1532634712/12e22b7fb72f82392b6d99170ca0be43", uri="sip:XXX.XXX.XXX.XXX:5061", response="8d5b346ae6ed353a31ab747650c7b435", algorithm=MD5, cnonce="0a4f113b", opaque="1f03ae9b1e858ca6", qop=auth, nc=00000002
Max-Forwards: 70
User-Agent: Yealink SIP-T28P 2.73.0.50
Content-Length: 0


[2018-07-26 14:52:08] VERBOSE[16840] res_pjsip_logger.c: <--- Transmitting SIP response (333 bytes) to UDP:YYY.YYY.YYY.YYY:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.165:5061;rport=5061;received=YYY.YYY.YYY.YYY;branch=z9hG4bK3718025555
Call-ID: [email protected]
From: "Peter J" <sip:[email protected]>;tag=3161788746
To: <sip:[email protected]>;tag=4e31566c-4dcf-44fa-834d-03bb16cee9e0
CSeq: 3 BYE
Server: Asterisk PBX 13.22.0
Content-Length: 0


[2018-07-26 14:52:08] VERBOSE[22620][C-00000007] app.c: User hung up
[2018-07-26 14:52:08] VERBOSE[22620][C-00000007] app_macro.c: Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on 'PJSIP/1002-0000000e' in macro 'vm'
[2018-07-26 14:52:08] VERBOSE[22620][C-00000007] app_macro.c: Spawn extension (macro-exten-vm, s, 21) exited non-zero on 'PJSIP/1002-0000000e' in macro 'exten-vm'
[2018-07-26 14:52:08] VERBOSE[22620][C-00000007] pbx.c: Spawn extension (ext-local, 1008, 2) exited non-zero on 'PJSIP/1002-0000000e'
[2018-07-26 14:52:08] VERBOSE[22620][C-00000007] pbx.c: Executing [[email protected]:1] Macro("PJSIP/1002-0000000e", "hangupcall,") in new stack
[2018-07-26 14:52:08] VERBOSE[22620][C-00000007] pbx.c: Executing [[email protected]:1] GotoIf("PJSIP/1002-0000000e", "1?theend") in new stack
[2018-07-26 14:52:08] VERBOSE[22620][C-00000007] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2018-07-26 14:52:08] VERBOSE[22620][C-00000007] pbx.c: Executing [[email protected]:3] ExecIf("PJSIP/1002-0000000e", "0?Set(CDR(recordingfile)=)") in new stack
[2018-07-26 14:52:08] VERBOSE[22620][C-00000007] pbx.c: Executing [[email protected]:4] Hangup("PJSIP/1002-0000000e", "") in new stack
[2018-07-26 14:52:08] VERBOSE[22620][C-00000007] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'PJSIP/1002-0000000e' in macro 'hangupcall'
[2018-07-26 14:52:08] VERBOSE[22620][C-00000007] pbx.c: Spawn extension (ext-local, h, 1) exited non-zero on 'PJSIP/1002-0000000e'
[2018-07-26 14:52:08] VERBOSE[16840] res_pjsip_logger.c: <--- Transmitting SIP request (629 bytes) to UDP:YYY.YYY.YYY.YYY:1027 --->
NOTIFY sip:[email protected]:1027 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5061;rport;branch=z9hG4bKPj88867a4d-3909-4211-9557-8e54b732d0c6
From: <sip:[email protected]>;tag=0f7a1ad4-d3a4-414d-bd12-7a557323d989
To: <sip:[email protected]>
Contact: <sip:[email protected]:5061>
Call-ID: ee1b0b79-d09c-4a14-ae79-22be8980ca54
CSeq: 22526 NOTIFY
Subscription-State: terminated
Event: message-summary
Allow-Events: presence, dialog, message-summary, refer
Max-Forwards: 70
User-Agent: Asterisk PBX 13.22.0
Content-Type: application/simple-message-summary
Content-Length: 49

Messages-Waiting: yes
Voice-Message: 1/0 (0/0)

[2018-07-26 14:52:08] VERBOSE[14848] res_pjsip_logger.c: <--- Received SIP response (335 bytes) from UDP:YYY.YYY.YYY.YYY:1027 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5061;rport;branch=z9hG4bKPj88867a4d-3909-4211-9557-8e54b732d0c6
From: <sip:[email protected]>;tag=0f7a1ad4-d3a4-414d-bd12-7a557323d989
To: <sip:[email protected]>
Call-ID: ee1b0b79-d09c-4a14-ae79-22be8980ca54
Date: Thu, 26 Jul 2018 19:55:03 GMT
CSeq: 22526 NOTIFY
Content-Length: 0
 
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