Pastein: <--- SIP read from UDP:192.168.0.200:5060 ---> REGISTER sip:192.168.0.76 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.200:5060;rport;branch=z9hG4bKPjreNU.-jWDWmcjH2VyQST1qHNGKyAhIgq Max-Forwards: 70 From: <

Загрузка данных
<--- SIP read from UDP:192.168.0.200:5060 --->
REGISTER sip:192.168.0.76 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.200:5060;rport;branch=z9hG4bKPjreNU.-jWDWmcjH2VyQST1qHNGKyAhIgq
Max-Forwards: 70
From: <sip:1005@192.168.0.76>;tag=4uRb0l.hAoqsidEKMu4mky6zPJtbSJvw
To: <sip:1005@192.168.0.76>
Call-ID: CYTnolCFJtl5kfeQyq99buavSyFOyAiz
CSeq: 12107 REGISTER
Contact: <sip:1005@192.168.0.200:5060;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.0.200:5060 (no NAT)
Sending to 192.168.0.200:5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.0.200:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bKPjreNU.-jWDWmcjH2VyQST1qHNGKyAhIgq;received=192.168.0.200;rport=5060
From: <sip:1005@192.168.0.76>;tag=4uRb0l.hAoqsidEKMu4mky6zPJtbSJvw
To: <sip:1005@192.168.0.76>;tag=as6e6afd60
Call-ID: CYTnolCFJtl5kfeQyq99buavSyFOyAiz
CSeq: 12107 REGISTER
Server: Asterisk PBX 11.25.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="744708b9"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'CYTnolCFJtl5kfeQyq99buavSyFOyAiz' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:192.168.0.200:5060 --->
REGISTER sip:192.168.0.76 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.200:5060;rport;branch=z9hG4bKPjuHh0Qs9V66lcuYMiDKfBsnYmxeferyg2
Max-Forwards: 70
From: <sip:1005@192.168.0.76>;tag=4uRb0l.hAoqsidEKMu4mky6zPJtbSJvw
To: <sip:1005@192.168.0.76>
Call-ID: CYTnolCFJtl5kfeQyq99buavSyFOyAiz
CSeq: 12108 REGISTER
Contact: <sip:1005@192.168.0.200:5060;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Authorization: Digest username="1005", realm="asterisk", nonce="744708b9", uri="sip:192.168.0.76", response="2ffb625a9c70761a70a36a3c76fe93bd", algorithm=MD5
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.0.200:5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.0.200:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bKPjuHh0Qs9V66lcuYMiDKfBsnYmxeferyg2;received=192.168.0.200;rport=5060
From: <sip:1005@192.168.0.76>;tag=4uRb0l.hAoqsidEKMu4mky6zPJtbSJvw
To: <sip:1005@192.168.0.76>;tag=as6e6afd60
Call-ID: CYTnolCFJtl5kfeQyq99buavSyFOyAiz
CSeq: 12108 REGISTER
Server: Asterisk PBX 11.25.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 300
Contact: <sip:1005@192.168.0.200:5060;ob>;expires=300
Date: Fri, 25 Sep 2026 12:30:53 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'CYTnolCFJtl5kfeQyq99buavSyFOyAiz' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:192.168.0.200:5060 --->
INVITE sip:1002@192.168.0.76 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.200:5060;rport;branch=z9hG4bKPjfELaxUpyzrpCoP.9iU.v2NOuhOBH5ACD
Max-Forwards: 70
From: sip:1005@192.168.0.76;tag=VdyD6VZ4weSC7YX770khee42u82tGjw1
To: sip:1002@192.168.0.76
Contact: <sip:1005@192.168.0.200:5060;ob>
Call-ID: rpSWAx.FWWNRMLhxOccHVTiN7wnHQ2hj
CSeq: 31619 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, siprec, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length: 838
v=0
o=- 2208989697 2208989697 IN IP4 192.168.0.200
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4004 RTP/AVP 96 97 98 99 3 0 8 9 120 121 122
c=IN IP4 192.168.0.200
b=TIAS:64000
a=rtcp:4005 IN IP4 192.168.0.200
a=sendrecv
a=rtpmap:96 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:98 speex/32000
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:120 telephone-event/16000
a=fmtp:120 0-16
a=rtpmap:121 telephone-event/8000
a=fmtp:121 0-16
a=rtpmap:122 telephone-event/32000
a=fmtp:122 0-16
a=ssrc:1757017423 cname:1a178a7414a36db1
m=text 4006 RTP/AVP 100 98
c=IN IP4 192.168.0.200
a=rtcp:4007 IN IP4 192.168.0.200
a=sendrecv
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98
a=rtpmap:98 t140/1000
a=ssrc:624785345 cname:1a178a7414a36db1
<------------->
--- (14 headers 35 lines) ---
Sending to 192.168.0.200:5060 (no NAT)
Sending to 192.168.0.200:5060 (no NAT)
Using INVITE request as basis request - rpSWAx.FWWNRMLhxOccHVTiN7wnHQ2hj
Found peer '1005' for '1005' from 192.168.0.200:5060
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 99
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 120
Found RTP audio format 121
Found RTP audio format 122
Found audio description format speex for ID 96
Found audio description format speex for ID 97
Found audio description format speex for ID 98
Found audio description format iLBC for ID 99
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found unknown media description format telephone-event for ID 120
Found audio description format telephone-event for ID 121
Found unknown media description format telephone-event for ID 122
Found RTP text format 100
Found RTP text format 98
Capabilities: us - (g723|gsm|ulaw|alaw|g729|speex|speex16|ilbc|g722|h264|speex32), peer - audio=(gsm|ulaw|alaw|speex|speex16|ilbc|g722|speex32)/video=(nothing)/text=(h263p|silk12), combined - (gsm|ulaw|alaw|speex|speex16|ilbc|g722|speex32)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.200:4004
Peer doesn't provide video
Looking for 1002 in outcoling (domain 192.168.0.76)
list_route: hop: <sip:1005@192.168.0.200:5060;ob>
<--- Transmitting (no NAT) to 192.168.0.200:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bKPjfELaxUpyzrpCoP.9iU.v2NOuhOBH5ACD;received=192.168.0.200;rport=5060
From: sip:1005@192.168.0.76;tag=VdyD6VZ4weSC7YX770khee42u82tGjw1
To: sip:1002@192.168.0.76
Call-ID: rpSWAx.FWWNRMLhxOccHVTiN7wnHQ2hj
CSeq: 31619 INVITE
Server: Asterisk PBX 11.25.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1002@192.168.0.76:5060>
Content-Length: 0
<------------>
    -- Executing [1002@outcoling:1] Dial("SIP/1005-00000008", "SIP/1002,,m") in new stack
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
Audio is at 16942
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100010 (ilbc) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 100001 (g723) to SDP
Adding codec 100009 (speex) to SDP
Adding codec 100016 (speex16) to SDP
Adding codec 100028 (speex32) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.5:5060:
INVITE sip:1002@192.168.0.5;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK7358f018
Max-Forwards: 70
From: <sip:1005@192.168.0.76>;tag=as3599431d
To: <sip:1002@192.168.0.5;transport=udp>
Contact: <sip:1005@192.168.0.76:5060>
Call-ID: 313261662dc077b02edc8d357ee2733a@192.168.0.76:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.25.3
Date: Fri, 25 Sep 2026 12:30:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 507
v=0
o=root 1650779050 1650779050 IN IP4 192.168.0.76
s=Asterisk PBX 11.25.3
c=IN IP4 192.168.0.76
t=0 0
m=audio 16942 RTP/AVP 8 0 3 99 18 9 4 97 96 98 121
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 iLBC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:97 speex/8000
a=rtpmap:96 speex/16000
a=rtpmap:98 speex/32000
a=rtpmap:121 telephone-event/8000
a=fmtp:121 0-16
a=ptime:20
a=sendrecv
---
    -- Called SIP/1002
    -- Music class default requested but no musiconhold loaded.
Audio is at 13660
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100010 (ilbc) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 100009 (speex) to SDP
Adding codec 100016 (speex16) to SDP
Adding codec 100028 (speex32) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (no NAT) to 192.168.0.200:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bKPjfELaxUpyzrpCoP.9iU.v2NOuhOBH5ACD;received=192.168.0.200;rport=5060
From: sip:1005@192.168.0.76;tag=VdyD6VZ4weSC7YX770khee42u82tGjw1
To: sip:1002@192.168.0.76;tag=as58cfb39c
Call-ID: rpSWAx.FWWNRMLhxOccHVTiN7wnHQ2hj
CSeq: 31619 INVITE
Server: Asterisk PBX 11.25.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1002@192.168.0.76:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 441
v=0
o=root 1315520761 1315520761 IN IP4 192.168.0.76
s=Asterisk PBX 11.25.3
c=IN IP4 192.168.0.76
t=0 0
m=audio 13660 RTP/AVP 8 0 3 99 9 97 96 98 121
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 iLBC/8000
a=rtpmap:9 G722/8000
a=rtpmap:97 speex/8000
a=rtpmap:96 speex/16000
a=rtpmap:98 speex/32000
a=rtpmap:121 telephone-event/8000
a=fmtp:121 0-16
a=ptime:20
a=sendrecv
m=text 0 RTP/AVP 100 98
<------------>
<--- SIP read from UDP:192.168.0.5:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK7358f018
From: <sip:1005@192.168.0.76>;tag=as3599431d
To: <sip:1002@192.168.0.5;transport=udp>
Call-ID: 313261662dc077b02edc8d357ee2733a@192.168.0.76:5060
CSeq: 102 INVITE
<------------->
--- (6 headers 0 lines) ---
<--- SIP read from UDP:192.168.0.5:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK7358f018
From: <sip:1005@192.168.0.76>;tag=as3599431d
To: <sip:1002@192.168.0.5;transport=udp>;tag=-4-rVXl
Call-ID: 313261662dc077b02edc8d357ee2733a@192.168.0.76:5060
CSeq: 102 INVITE
User-Agent: Linphone-Desktop/6.1.1 (DESKTOP-SYA2882) windows/10 Qt/6.9.1 LinphoneSDK/5.4.88
Supported: replaces, outbound, gruu, path, record-aware
<------------->
--- (8 headers 0 lines) ---
list_route: no route
    -- SIP/1002-00000009 is ringing
<--- SIP read from UDP:192.168.0.5:5060 --->
<------------->
<--- SIP read from UDP:192.168.0.200:5060 --->
CANCEL sip:1002@192.168.0.76 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.200:5060;rport;branch=z9hG4bKPjfELaxUpyzrpCoP.9iU.v2NOuhOBH5ACD
Max-Forwards: 70
From: sip:1005@192.168.0.76;tag=VdyD6VZ4weSC7YX770khee42u82tGjw1
To: sip:1002@192.168.0.76
Call-ID: rpSWAx.FWWNRMLhxOccHVTiN7wnHQ2hj
CSeq: 31619 CANCEL
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Sending to 192.168.0.200:5060 (no NAT)
<--- Reliably Transmitting (no NAT) to 192.168.0.200:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bKPjfELaxUpyzrpCoP.9iU.v2NOuhOBH5ACD;received=192.168.0.200;rport=5060
From: sip:1005@192.168.0.76;tag=VdyD6VZ4weSC7YX770khee42u82tGjw1
To: sip:1002@192.168.0.76;tag=as58cfb39c
Call-ID: rpSWAx.FWWNRMLhxOccHVTiN7wnHQ2hj
CSeq: 31619 INVITE
Server: Asterisk PBX 11.25.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 192.168.0.200:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bKPjfELaxUpyzrpCoP.9iU.v2NOuhOBH5ACD;received=192.168.0.200;rport=5060
From: sip:1005@192.168.0.76;tag=VdyD6VZ4weSC7YX770khee42u82tGjw1
To: sip:1002@192.168.0.76;tag=as58cfb39c
Call-ID: rpSWAx.FWWNRMLhxOccHVTiN7wnHQ2hj
CSeq: 31619 CANCEL
Server: Asterisk PBX 11.25.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '313261662dc077b02edc8d357ee2733a@192.168.0.76:5060' in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.0.5:5060:
CANCEL sip:1002@192.168.0.5;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK7358f018
Max-Forwards: 70
From: <sip:1005@192.168.0.76>;tag=as3599431d
To: <sip:1002@192.168.0.5;transport=udp>
Call-ID: 313261662dc077b02edc8d357ee2733a@192.168.0.76:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 11.25.3
Content-Length: 0
---
Scheduling destruction of SIP dialog '313261662dc077b02edc8d357ee2733a@192.168.0.76:5060' in 32000 ms (Method: INVITE)
  == Spawn extension (outcoling, 1002, 1) exited non-zero on 'SIP/1005-00000008'
<--- SIP read from UDP:192.168.0.200:5060 --->
ACK sip:1002@192.168.0.76 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.200:5060;rport;branch=z9hG4bKPjfELaxUpyzrpCoP.9iU.v2NOuhOBH5ACD
Max-Forwards: 70
From: sip:1005@192.168.0.76;tag=VdyD6VZ4weSC7YX770khee42u82tGjw1
To: sip:1002@192.168.0.76;tag=as58cfb39c
Call-ID: rpSWAx.FWWNRMLhxOccHVTiN7wnHQ2hj
CSeq: 31619 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'rpSWAx.FWWNRMLhxOccHVTiN7wnHQ2hj' Method: ACK
<--- SIP read from UDP:192.168.0.5:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK7358f018
From: <sip:1005@192.168.0.76>;tag=as3599431d
To: <sip:1002@192.168.0.5;transport=udp>;tag=gMSOZFf
Call-ID: 313261662dc077b02edc8d357ee2733a@192.168.0.76:5060
CSeq: 102 CANCEL
User-Agent: Linphone-Desktop/6.1.1 (DESKTOP-SYA2882) windows/10 Qt/6.9.1 LinphoneSDK/5.4.88
Supported: replaces, outbound, gruu, path, record-aware
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:192.168.0.5:5060 --->
SIP/2.0 487 Request terminated
Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK7358f018
From: <sip:1005@192.168.0.76>;tag=as3599431d
To: <sip:1002@192.168.0.5;transport=udp>;tag=-4-rVXl
Call-ID: 313261662dc077b02edc8d357ee2733a@192.168.0.76:5060
CSeq: 102 INVITE
User-Agent: Linphone-Desktop/6.1.1 (DESKTOP-SYA2882) windows/10 Qt/6.9.1 LinphoneSDK/5.4.88
Supported: replaces, outbound, gruu, path, record-aware
<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 192.168.0.5:5060:
ACK sip:1002@192.168.0.5;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK7358f018
Max-Forwards: 70
From: <sip:1005@192.168.0.76>;tag=as3599431d
To: <sip:1002@192.168.0.5;transport=udp>;tag=-4-rVXl
Contact: <sip:1005@192.168.0.76:5060>
Call-ID: 313261662dc077b02edc8d357ee2733a@192.168.0.76:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.25.3
Content-Length: 0
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