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Debug log
From: "101"<sip:101@192.168.1.10>;tag=f8860cc5
To: "08000556688"<sip:08000556688@192.168.1.10>;tag=as 6e4cb0ae
Call-ID: MjMxNTgxMTViOThmNDliOTg3ZGIwMDRhNGZkMTU2MDI.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:08000556688@192.168.1.10>
Content-Type: application/sdp
Content-Length: 259
v=0
o=root 699117025 699117025 IN IP4 192.168.1.10
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.1.10
t=0 0
m=audio 10684 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
-- Executing [08000556688@from-internal:4] Wait("SIP/101-00000007", "1") in new stack
-- Executing [08000556688@from-internal:5] Playback("SIP/101-00000007", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
-- <SIP/101-00000007> Playing 'silence/1.gsm' (language 'en')
-- <SIP/101-00000007> Playing 'cannot-complete-as-dialed.gsm' (language 'en')
<--- SIP read from UDP:192.168.1.7:5940 --->
<------------->
<--- SIP read from UDP:192.168.1.7:5940 --->
SUBSCRIBE sip:asterisk@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5940;branch=z9hG4bK-d8754z-abecc7ceeb82160b-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:101@192.168.1.7:5940>
To: "101"<sip:101@192.168.1.10>;tag=as68828cc3
From: "101"<sip:101@192.168.1.10>;tag=17d984e5
Call-ID: NjgzNDBhMTQyMzFiMWI5ZGQ2OGRmMzJmZjk4YTY5Nzc.
CSeq: 19 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite 4 release 4.0 stamp 58832
Authorization: Digest username="101",realm="asterisk",nonce="76a7e4d8",u ri="sip:asterisk@192.168.1.10",response="7b612db84 bcef48a394e454893379f1d",algorithm=MD5
Event: message-summary
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Found peer '101' for '101' from 192.168.1.7:5940
<--- Transmitting (NAT) to 192.168.1.7:5940 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.7:5940;branch=z9hG4bK-d8754z-abecc7ceeb82160b-1---d8754z-;received=192.168.1.7;rport=5940
From: "101"<sip:101@192.168.1.10>;tag=17d984e5
To: "101"<sip:101@192.168.1.10>;tag=as68828cc3
Call-ID: NjgzNDBhMTQyMzFiMWI5ZGQ2OGRmMzJmZjk4YTY5Nzc.
CSeq: 19 SUBSCRIBE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="590d3888", stale=true
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'NjgzNDBhMTQyMzFiMWI5ZGQ2OGRmMzJmZjk4YTY5Nzc.' in 6400 ms (Method: SUBSCRIBE)
<--- SIP read from UDP:192.168.1.7:5940 --->
SUBSCRIBE sip:asterisk@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5940;branch=z9hG4bK-d8754z-bbfcc56806814850-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:101@192.168.1.7:5940>
To: "101"<sip:101@192.168.1.10>;tag=as68828cc3
From: "101"<sip:101@192.168.1.10>;tag=17d984e5
Call-ID: NjgzNDBhMTQyMzFiMWI5ZGQ2OGRmMzJmZjk4YTY5Nzc.
CSeq: 20 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite 4 release 4.0 stamp 58832
Authorization: Digest username="101",realm="asterisk",nonce="590d3888",u ri="sip:asterisk@192.168.1.10",response="a93378aeb 1943589c56d8c8b783d64fb",algorithm=MD5
Event: message-summary
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Found peer '101' for '101' from 192.168.1.7:5940
Scheduling destruction of SIP dialog 'NjgzNDBhMTQyMzFiMWI5ZGQ2OGRmMzJmZjk4YTY5Nzc.' in 310000 ms (Method: SUBSCRIBE)
<--- Transmitting (NAT) to 192.168.1.7:5940 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.7:5940;branch=z9hG4bK-d8754z-bbfcc56806814850-1---d8754z-;received=192.168.1.7;rport=5940
From: "101"<sip:101@192.168.1.10>;tag=17d984e5
To: "101"<sip:101@192.168.1.10>;tag=as68828cc3
Call-ID: NjgzNDBhMTQyMzFiMWI5ZGQ2OGRmMzJmZjk4YTY5Nzc.
CSeq: 20 SUBSCRIBE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 300
Contact: <sip:asterisk@192.168.1.10>;expires=300
Content-Length: 0
<------------>
Reliably Transmitting (NAT) to 192.168.1.7:5940:
NOTIFY sip:101@192.168.1.7:5940 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK689d9512;rport
Max-Forwards: 70
Route: <sip:101@192.168.1.7:5940>
From: "asterisk" <sip:asterisk@192.168.1.10>;tag=as68828cc3
To: <sip:101@192.168.1.7:5940>;tag=17d984e5
Contact: <sip:asterisk@192.168.1.10>
Call-ID: NjgzNDBhMTQyMzFiMWI5ZGQ2OGRmMzJmZjk4YTY5Nzc.
CSeq: 111 NOTIFY
User-Agent: Asterisk PBX 1.6.2.13
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 92
Messages-Waiting: no
Message-Account: sip:asterisk@192.168.1.10
Voice-Message: 0/0 (0/0)
---
<--- SIP read from UDP:192.168.1.7:5940 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK689d9512;rport=506 0
Contact: <sip:101@192.168.1.7:5940>
To: <sip:101@192.168.1.7:5940>;tag=17d984e5
From: "asterisk"<sip:asterisk@192.168.1.10>;tag=as68828c c3
Call-ID: NjgzNDBhMTQyMzFiMWI5ZGQ2OGRmMzJmZjk4YTY5Nzc.
CSeq: 111 NOTIFY
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
-- <SIP/101-00000007> Playing 'check-number-dial-again.gsm' (language 'en')
-- Executing [08000556688@from-internal:6] Wait("SIP/101-00000007", "1") in new stack
-- Executing [08000556688@from-internal:7] Congestion("SIP/101-00000007", "20") in new stack
<--- Reliably Transmitting (NAT) to 192.168.1.7:5940 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.7:5940;branch=z9hG4bK-d8754z-3df9eb35f1cad59c-1---d8754z-;received=192.168.1.7;rport=5940
From: "101"<sip:101@192.168.1.10>;tag=f8860cc5
To: "08000556688"<sip:08000556688@192.168.1.10>;tag=as 6e4cb0ae
Call-ID: MjMxNTgxMTViOThmNDliOTg3ZGIwMDRhNGZkMTU2MDI.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
== Spawn extension (from-internal, 08000556688, 7) exited non-zero on 'SIP/101-00000007'
-- Executing [h@from-internal:1] Macro("SIP/101-00000007", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/101-00000007", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] NoOp("SIP/101-00000007", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/101-00000007", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/101-00000007", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,10)
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/101-00000007", "1?theend") in new stack
-- Goto (macro-hangupcall,s,12)
-- Executing [s@macro-hangupcall:12] Hangup("SIP/101-00000007", "") in new stack
== Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/101-00000007' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/101-00000007'
<--- SIP read from UDP:192.168.1.7:5940 --->
ACK sip:08000556688@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.7:5940;branch=z9hG4bK-d8754z-3df9eb35f1cad59c-1---d8754z-;rport
Max-Forwards: 70
To: "08000556688"<sip:08000556688@192.168.1.10>;tag=as 6e4cb0ae
From: "101"<sip:101@192.168.1.10>;tag=f8860cc5
Call-ID: MjMxNTgxMTViOThmNDliOTg3ZGIwMDRhNGZkMTU2MDI.
CSeq: 2 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'MjMxNTgxMTViOThmNDliOTg3ZGIwMDRhNGZkMTU2MDI.' Method: ACK
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