Advertisement
Not a member of Pastebin yet?
Sign Up,
it unlocks many cool features!
- [Jan 16 14:41:48] <--- SIP read from UDP:192.168.0.52:5060 --->
- [Jan 16 14:41:48] INVITE sip:[email protected] SIP/2.0
- [Jan 16 14:41:48] Via: SIP/2.0/UDP 192.168.0.52:5060;branch=z9hG4bK-2czztm13kjqx;rport
- [Jan 16 14:41:48] From: "Buero" <sip:[email protected]>;tag=4dl93g4h84
- [Jan 16 14:41:48] To: <sip:[email protected]>
- [Jan 16 14:41:48] Call-ID: 54b915195f41-tvx8stu7cwdi
- [Jan 16 14:41:48] CSeq: 1 INVITE
- [Jan 16 14:41:48] Max-Forwards: 70
- [Jan 16 14:41:48] Contact: <sip:[email protected]:5060>;reg-id=1
- [Jan 16 14:41:48] X-Serialnumber: 0004132963A8
- [Jan 16 14:41:48] P-Key-Flags: resolution="31x13", keys="4"
- [Jan 16 14:41:48] User-Agent: snom360/8.7.3.25
- [Jan 16 14:41:48] Accept: application/sdp
- [Jan 16 14:41:48] Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
- [Jan 16 14:41:48] Allow-Events: talk, hold, refer, call-info
- [Jan 16 14:41:48] Supported: timer, 100rel, replaces, from-change
- [Jan 16 14:41:48] Session-Expires: 3600;refresher=uas
- [Jan 16 14:41:48] Min-SE: 90
- [Jan 16 14:41:48] Content-Type: application/sdp
- [Jan 16 14:41:48] Content-Length: 487
- [Jan 16 14:41:48]
- [Jan 16 14:41:48] v=0
- [Jan 16 14:41:48] o=root 2014030264 2014030264 IN IP4 192.168.0.52
- [Jan 16 14:41:48] s=call
- [Jan 16 14:41:48] c=IN IP4 192.168.0.52
- [Jan 16 14:41:48] t=0 0
- [Jan 16 14:41:48] m=audio 56520 RTP/AVP 9 0 8 3 99 108 18 101
- [Jan 16 14:41:48] a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:PvtxmlqGatT4oUwvqUhDwt79elOhiIx1oZa5hQr+
- [Jan 16 14:41:48] a=rtpmap:9 G722/8000
- [Jan 16 14:41:48] a=rtpmap:0 PCMU/8000
- [Jan 16 14:41:48] a=rtpmap:8 PCMA/8000
- [Jan 16 14:41:48] a=rtpmap:3 GSM/8000
- [Jan 16 14:41:48] a=rtpmap:99 G726-32/8000
- [Jan 16 14:41:48] a=rtpmap:108 AAL2-G726-32/8000
- [Jan 16 14:41:48] a=rtpmap:18 G729/8000
- [Jan 16 14:41:48] a=fmtp:18 annexb=no
- [Jan 16 14:41:48] a=rtpmap:101 telephone-event/8000
- [Jan 16 14:41:48] a=fmtp:101 0-15
- [Jan 16 14:41:48] a=ptime:20
- [Jan 16 14:41:48] a=sendrecv
- [Jan 16 14:41:48] <------------->
- [Jan 16 14:41:48] --- (19 headers 19 lines) ---
- [Jan 16 14:41:48] Sending to 192.168.0.52:5060 (no NAT)
- [Jan 16 14:41:48] Sending to 192.168.0.52:5060 (no NAT)
- [Jan 16 14:41:48] Using INVITE request as basis request - 54b915195f41-tvx8stu7cwdi
- [Jan 16 14:41:48] Found peer '91' for '91' from 192.168.0.52:5060
- [Jan 16 14:41:48]
- [Jan 16 14:41:48] <--- Reliably Transmitting (no NAT) to 192.168.0.52:5060 --->
- [Jan 16 14:41:48] SIP/2.0 401 Unauthorized
- [Jan 16 14:41:48] Via: SIP/2.0/UDP 192.168.0.52:5060;branch=z9hG4bK-2czztm13kjqx;received=192.168.0.52;rport=5060
- [Jan 16 14:41:48] From: "Buero" <sip:[email protected]>;tag=4dl93g4h84
- [Jan 16 14:41:48] To: <sip:[email protected]>;tag=as00451302
- [Jan 16 14:41:48] Call-ID: 54b915195f41-tvx8stu7cwdi
- [Jan 16 14:41:48] CSeq: 1 INVITE
- [Jan 16 14:41:48] Server: Asterisk PBX 13.1.0
- [Jan 16 14:41:48] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- [Jan 16 14:41:48] Supported: replaces, timer
- [Jan 16 14:41:48] WWW-Authenticate: Digest algorithm=MD5, realm="kaworu.kunbox.net", nonce="75387cce"
- [Jan 16 14:41:48] Content-Length: 0
- [Jan 16 14:41:48]
- [Jan 16 14:41:48]
- [Jan 16 14:41:48] <------------>
- [Jan 16 14:41:48] Scheduling destruction of SIP dialog '54b915195f41-tvx8stu7cwdi' in 6400 ms (Method: INVITE)
- [Jan 16 14:41:48]
- [Jan 16 14:41:48] <--- SIP read from UDP:192.168.0.150:21494 --->
- [Jan 16 14:41:48] SIP/2.0 200 OK
- [Jan 16 14:41:48] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK4b024d96
- [Jan 16 14:41:48] Contact: <sip:192.168.0.150:21494>
- [Jan 16 14:41:48] To: <sip:[email protected]:21494;rinstance=2d7e645d3613516b>;tag=23df8963
- [Jan 16 14:41:48] From: "asterisk"<sip:[email protected]>;tag=as2d6fae3d
- [Jan 16 14:41:48] Call-ID: [email protected]:5060
- [Jan 16 14:41:48] CSeq: 102 OPTIONS
- [Jan 16 14:41:48] Accept: application/sdp
- [Jan 16 14:41:48] Accept-Language: en
- [Jan 16 14:41:48] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
- [Jan 16 14:41:48] Supported: replaces, eventlist
- [Jan 16 14:41:48] User-Agent: X-Lite release 4.7.0 stamp 73588 e332378a-M
- [Jan 16 14:41:48] Content-Length: 0
- [Jan 16 14:41:48]
- [Jan 16 14:41:48] <------------->
- [Jan 16 14:41:48] --- (13 headers 0 lines) ---
- [Jan 16 14:41:48] Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
- [Jan 16 14:41:48]
- [Jan 16 14:41:48] <--- SIP read from UDP:192.168.0.52:5060 --->
- [Jan 16 14:41:48] ACK sip:[email protected] SIP/2.0
- [Jan 16 14:41:48] Via: SIP/2.0/UDP 192.168.0.52:5060;branch=z9hG4bK-2czztm13kjqx;rport
- [Jan 16 14:41:48] From: "Buero" <sip:[email protected]>;tag=4dl93g4h84
- [Jan 16 14:41:48] To: <sip:[email protected]>;tag=as00451302
- [Jan 16 14:41:48] Call-ID: 54b915195f41-tvx8stu7cwdi
- [Jan 16 14:41:48] CSeq: 1 ACK
- [Jan 16 14:41:48] Max-Forwards: 70
- [Jan 16 14:41:48] Contact: <sip:[email protected]:5060>;reg-id=1
- [Jan 16 14:41:48] Content-Length: 0
- [Jan 16 14:41:48]
- [Jan 16 14:41:48] <------------->
- [Jan 16 14:41:48] --- (9 headers 0 lines) ---
- [Jan 16 14:41:48]
- [Jan 16 14:41:48] <--- SIP read from UDP:192.168.0.52:5060 --->
- [Jan 16 14:41:48] INVITE sip:[email protected] SIP/2.0
- [Jan 16 14:41:48] Via: SIP/2.0/UDP 192.168.0.52:5060;branch=z9hG4bK-62pgfuna2gj0;rport
- [Jan 16 14:41:48] From: "Buero" <sip:[email protected]>;tag=4dl93g4h84
- [Jan 16 14:41:48] To: <sip:[email protected]>
- [Jan 16 14:41:48] Call-ID: 54b915195f41-tvx8stu7cwdi
- [Jan 16 14:41:48] CSeq: 2 INVITE
- [Jan 16 14:41:48] Max-Forwards: 70
- [Jan 16 14:41:48] Contact: <sip:[email protected]:5060>;reg-id=1
- [Jan 16 14:41:48] X-Serialnumber: 0004132963A8
- [Jan 16 14:41:48] P-Key-Flags: resolution="31x13", keys="4"
- [Jan 16 14:41:48] User-Agent: snom360/8.7.3.25
- [Jan 16 14:41:48] Accept: application/sdp
- [Jan 16 14:41:48] Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
- [Jan 16 14:41:48] Allow-Events: talk, hold, refer, call-info
- [Jan 16 14:41:48] Supported: timer, 100rel, replaces, from-change
- [Jan 16 14:41:48] Session-Expires: 3600;refresher=uas
- [Jan 16 14:41:48] Min-SE: 90
- [Jan 16 14:41:48] Authorization: Digest username="91",realm="kaworu.kunbox.net",nonce="75387cce",uri="sip:[email protected]",response="2601f318c80699e611417802c651356b",algorithm=MD5
- [Jan 16 14:41:48] Content-Type: application/sdp
- [Jan 16 14:41:48] Content-Length: 487
- [Jan 16 14:41:48]
- [Jan 16 14:41:48] v=0
- [Jan 16 14:41:48] o=root 2014030264 2014030264 IN IP4 192.168.0.52
- [Jan 16 14:41:48] s=call
- [Jan 16 14:41:48] c=IN IP4 192.168.0.52
- [Jan 16 14:41:48] t=0 0
- [Jan 16 14:41:48] m=audio 56520 RTP/AVP 9 0 8 3 99 108 18 101
- [Jan 16 14:41:48] a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:PvtxmlqGatT4oUwvqUhDwt79elOhiIx1oZa5hQr+
- [Jan 16 14:41:48] a=rtpmap:9 G722/8000
- [Jan 16 14:41:48] a=rtpmap:0 PCMU/8000
- [Jan 16 14:41:48] a=rtpmap:8 PCMA/8000
- [Jan 16 14:41:48] a=rtpmap:3 GSM/8000
- [Jan 16 14:41:48] a=rtpmap:99 G726-32/8000
- [Jan 16 14:41:48] a=rtpmap:108 AAL2-G726-32/8000
- [Jan 16 14:41:48] a=rtpmap:18 G729/8000
- [Jan 16 14:41:48] a=fmtp:18 annexb=no
- [Jan 16 14:41:48] a=rtpmap:101 telephone-event/8000
- [Jan 16 14:41:48] a=fmtp:101 0-15
- [Jan 16 14:41:48] a=ptime:20
- [Jan 16 14:41:48] a=sendrecv
- [Jan 16 14:41:48] <------------->
- [Jan 16 14:41:48] --- (20 headers 19 lines) ---
- [Jan 16 14:41:48] Sending to 192.168.0.52:5060 (no NAT)
- [Jan 16 14:41:48] Using INVITE request as basis request - 54b915195f41-tvx8stu7cwdi
- [Jan 16 14:41:48] Found peer '91' for '91' from 192.168.0.52:5060
- [Jan 16 14:41:48] == Using SIP RTP CoS mark 5
- [Jan 16 14:41:48] Found RTP audio format 9
- [Jan 16 14:41:48] Found RTP audio format 0
- [Jan 16 14:41:48] Found RTP audio format 8
- [Jan 16 14:41:48] Found RTP audio format 3
- [Jan 16 14:41:48] Found RTP audio format 99
- [Jan 16 14:41:48] Found RTP audio format 108
- [Jan 16 14:41:48] Found RTP audio format 18
- [Jan 16 14:41:48] Found RTP audio format 101
- [Jan 16 14:41:48] Found audio description format G722 for ID 9
- [Jan 16 14:41:48] Found audio description format PCMU for ID 0
- [Jan 16 14:41:48] Found audio description format PCMA for ID 8
- [Jan 16 14:41:48] Found audio description format GSM for ID 3
- [Jan 16 14:41:48] Found audio description format G726-32 for ID 99
- [Jan 16 14:41:48] Found audio description format AAL2-G726-32 for ID 108
- [Jan 16 14:41:48] Found audio description format G729 for ID 18
- [Jan 16 14:41:48] Found audio description format telephone-event for ID 101
- [Jan 16 14:41:48] failed to extend from 64 to 98
- [Jan 16 14:41:48] Capabilities: us - (ulaw|alaw|gsm|h263|g723|g726|g726aal2|adpcm|slin|slin|slin|), peer - audio=(ulaw|gsm|alaw|g722|g729|g726|g726aal2)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm|g726|g726aal2|g729|g722)
- [Jan 16 14:41:48] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
- [Jan 16 14:41:48] Peer audio RTP is at port 192.168.0.52:56520
- [Jan 16 14:41:48] Looking for 98 in internal (domain 192.168.0.20)
- [Jan 16 14:41:48] set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
- [Jan 16 14:41:48] set_destination: set destination to 192.168.0.51:5060
- [Jan 16 14:41:48] Reliably Transmitting (no NAT) to 192.168.0.51:5060:
- [Jan 16 14:41:48] NOTIFY sip:[email protected]:5060 SIP/2.0
- [Jan 16 14:41:48] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK6cf6199c;rport
- [Jan 16 14:41:48] Max-Forwards: 70
- [Jan 16 14:41:48] From: <sip:[email protected]>;tag=as3be3a100
- [Jan 16 14:41:48] To: <sip:[email protected]>;tag=yqam73tqu3
- [Jan 16 14:41:48] Contact: <sip:[email protected]:5060>
- [Jan 16 14:41:48] Call-ID: 386d43a9e00b-xh9vubfgjat8
- [Jan 16 14:41:48] CSeq: 180 NOTIFY
- [Jan 16 14:41:48] User-Agent: Asterisk PBX 13.1.0
- [Jan 16 14:41:48] Subscription-State: active
- [Jan 16 14:41:48] Event: dialog
- [Jan 16 14:41:48] Content-Type: application/dialog-info+xml
- [Jan 16 14:41:48] Content-Length: 201
- [Jan 16 14:41:48]
- [Jan 16 14:41:48] <?xml version="1.0"?>
- [Jan 16 14:41:48] <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="78" state="full" entity="sip:[email protected]">
- [Jan 16 14:41:48] <dialog id="91">
- [Jan 16 14:41:48] <state>confirmed</state>
- [Jan 16 14:41:48] </dialog>
- [Jan 16 14:41:48] </dialog-info>
- [Jan 16 14:41:48]
- [Jan 16 14:41:48] ---
- [Jan 16 14:41:48] == Extension Changed 91[SoftPhone] new state InUse for Notify User 76
- [Jan 16 14:41:48] set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
- [Jan 16 14:41:48] set_destination: set destination to 192.168.0.52:5060
- [Jan 16 14:41:48] Reliably Transmitting (no NAT) to 192.168.0.52:5060:
- [Jan 16 14:41:48] NOTIFY sip:[email protected]:5060 SIP/2.0
- [Jan 16 14:41:48] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK3f8e0101;rport
- [Jan 16 14:41:48] Max-Forwards: 70
- [Jan 16 14:41:48] From: <sip:[email protected]>;tag=as45326187
- [Jan 16 14:41:48] To: <sip:[email protected]>;tag=fv4lqpihz2
- [Jan 16 14:41:48] Contact: <sip:[email protected]:5060>
- [Jan 16 14:41:48] Call-ID: 54b910bee022-dgmsn8yhawk1
- [Jan 16 14:41:48] CSeq: 117 NOTIFY
- [Jan 16 14:41:48] User-Agent: Asterisk PBX 13.1.0
- [Jan 16 14:41:48] Subscription-State: active
- [Jan 16 14:41:48] Event: dialog
- [Jan 16 14:41:48] Content-Type: application/dialog-info+xml
- [Jan 16 14:41:48] Content-Length: 201
- [Jan 16 14:41:48]
- [Jan 16 14:41:48] <?xml version="1.0"?>
- [Jan 16 14:41:48] <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="15" state="full" entity="sip:[email protected]">
- [Jan 16 14:41:48] <dialog id="91">
- [Jan 16 14:41:48] <state>confirmed</state>
- [Jan 16 14:41:48] </dialog>
- [Jan 16 14:41:48] </dialog-info>
- [Jan 16 14:41:48]
- [Jan 16 14:41:48] ---
- [Jan 16 14:41:48] == Extension Changed 91[SoftPhone] new state InUse for Notify User 91
- [Jan 16 14:41:48] sip_route_dump: route/path hop: <sip:[email protected]:5060>
- [Jan 16 14:41:48]
- [Jan 16 14:41:48] <--- Transmitting (no NAT) to 192.168.0.52:5060 --->
- [Jan 16 14:41:48] SIP/2.0 100 Trying
- [Jan 16 14:41:48] Via: SIP/2.0/UDP 192.168.0.52:5060;branch=z9hG4bK-62pgfuna2gj0;received=192.168.0.52;rport=5060
- [Jan 16 14:41:48] From: "Buero" <sip:[email protected]>;tag=4dl93g4h84
- [Jan 16 14:41:48] To: <sip:[email protected]>
- [Jan 16 14:41:48] Call-ID: 54b915195f41-tvx8stu7cwdi
- [Jan 16 14:41:48] CSeq: 2 INVITE
- [Jan 16 14:41:48] Server: Asterisk PBX 13.1.0
- [Jan 16 14:41:48] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- [Jan 16 14:41:48] Supported: replaces, timer
- [Jan 16 14:41:48] Session-Expires: 1800;refresher=uas
- [Jan 16 14:41:48] Contact: <sip:[email protected]:5060>
- [Jan 16 14:41:48] Content-Length: 0
- [Jan 16 14:41:48]
- [Jan 16 14:41:48]
- [Jan 16 14:41:48] <------------>
- [Jan 16 14:41:48] -- Executing [98@internal:1] VoiceMailMain("SIP/91-00000063", "800,s") in new stack
- [Jan 16 14:41:48] Audio is at 10700
- [Jan 16 14:41:48] Adding codec ulaw to SDP
- [Jan 16 14:41:48] Adding codec alaw to SDP
- [Jan 16 14:41:48] Adding codec gsm to SDP
- [Jan 16 14:41:48] Adding codec g726 to SDP
- [Jan 16 14:41:48] Adding codec g726aal2 to SDP
- [Jan 16 14:41:48] Adding codec g729 to SDP
- [Jan 16 14:41:48] Adding codec g722 to SDP
- [Jan 16 14:41:48] Adding codec g723 to SDP
- [Jan 16 14:41:48] Adding codec adpcm to SDP
- [Jan 16 14:41:48] Adding codec slin to SDP
- [Jan 16 14:41:48] Adding codec slin to SDP
- [Jan 16 14:41:48] Adding codec slin to SDP
- [Jan 16 14:41:48] Adding codec slin to SDP
- [Jan 16 14:41:48] Adding codec slin to SDP
- [Jan 16 14:41:48] Adding codec slin to SDP
- [Jan 16 14:41:48] Adding codec slin to SDP
- [Jan 16 14:41:48] Adding codec slin to SDP
- [Jan 16 14:41:48] Adding codec slin to SDP
- [Jan 16 14:41:48] Adding codec lpc10 to SDP
- [Jan 16 14:41:48] Adding codec speex to SDP
- [Jan 16 14:41:48] Adding codec speex to SDP
- [Jan 16 14:41:48] Adding codec speex to SDP
- [Jan 16 14:41:48] Adding codec ilbc to SDP
- [Jan 16 14:41:48] Adding codec siren7 to SDP
- [Jan 16 14:41:48] Adding codec siren14 to SDP
- [Jan 16 14:41:48] Adding codec testlaw to SDP
- [Jan 16 14:41:48] Adding codec g719 to SDP
- [Jan 16 14:41:48] Adding codec opus to SDP
- [Jan 16 14:41:48] Adding codec none to SDP
- [Jan 16 14:41:48]
- [Jan 16 14:41:48] <--- SIP read from UDP:192.168.0.51:5060 --->
- [Jan 16 14:41:48] SIP/2.0 200 Ok
- [Jan 16 14:41:48] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK6cf6199c;rport=5060
- [Jan 16 14:41:48] From: <sip:[email protected]>;tag=as3be3a100
- [Jan 16 14:41:48] To: <sip:[email protected]>;tag=yqam73tqu3
- [Jan 16 14:41:48] Call-ID: 386d43a9e00b-xh9vubfgjat8
- [Jan 16 14:41:48] CSeq: 180 NOTIFY
- [Jan 16 14:41:48] Content-Length: 0
- [Jan 16 14:41:48]
- [Jan 16 14:41:48] <------------->
- [Jan 16 14:41:48] --- (7 headers 0 lines) ---
- [Jan 16 14:41:48] Adding non-codec 0x1 (telephone-event) to SDP
- [Jan 16 14:41:48]
- [Jan 16 14:41:48] <--- Reliably Transmitting (no NAT) to 192.168.0.52:5060 --->
- [Jan 16 14:41:48] SIP/2.0 200 OK
- [Jan 16 14:41:48] Via: SIP/2.0/UDP 192.168.0.52:5060;branch=z9hG4bK-62pgfuna2gj0;received=192.168.0.52;rport=5060
- [Jan 16 14:41:48] From: "Buero" <sip:[email protected]>;tag=4dl93g4h84
- [Jan 16 14:41:48] To: <sip:[email protected]>;tag=as581dc455
- [Jan 16 14:41:48] Call-ID: 54b915195f41-tvx8stu7cwdi
- [Jan 16 14:41:48] CSeq: 2 INVITE
- [Jan 16 14:41:48] Server: Asterisk PBX 13.1.0
- [Jan 16 14:41:48] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- [Jan 16 14:41:48] Supported: replaces, timer
- [Jan 16 14:41:48] Session-Expires: 1800;refresher=uas
- [Jan 16 14:41:48] Contact: <sip:[email protected]:5060>
- [Jan 16 14:41:48] Content-Type: application/sdp
- [Jan 16 14:41:48] Require: timer
- [Jan 16 14:41:48] Content-Length: 896
- [Jan 16 14:41:48]
- [Jan 16 14:41:48] v=0
- [Jan 16 14:41:48] o=root 1979030603 1979030603 IN IP4 192.168.0.20
- [Jan 16 14:41:48] s=Asterisk PBX 13.1.0
- [Jan 16 14:41:48] c=IN IP4 192.168.0.20
- [Jan 16 14:41:48] t=0 0
- [Jan 16 14:41:48] m=audio 10700 RTP/AVP 0 8 3 99 108 18 9 4 5 10 118 7 110 117 119 97 102 115 116 107 101
- [Jan 16 14:41:48] a=rtpmap:0 PCMU/8000
- [Jan 16 14:41:48] a=rtpmap:8 PCMA/8000
- [Jan 16 14:41:48] a=rtpmap:3 GSM/8000
- [Jan 16 14:41:48] a=rtpmap:99 G726-32/8000
- [Jan 16 14:41:48] a=rtpmap:108 AAL2-G726-32/8000
- [Jan 16 14:41:48] a=rtpmap:18 G729/8000
- [Jan 16 14:41:48] a=fmtp:18 annexb=no
- [Jan 16 14:41:48] a=rtpmap:9 G722/8000
- [Jan 16 14:41:48] a=rtpmap:4 G723/8000
- [Jan 16 14:41:48] a=fmtp:4 annexa=no
- [Jan 16 14:41:48] a=rtpmap:5 DVI4/8000
- [Jan 16 14:41:48] a=rtpmap:10 L16/8000
- [Jan 16 14:41:48] a=rtpmap:118 L16/16000
- [Jan 16 14:41:48] a=rtpmap:7 LPC/8000
- [Jan 16 14:41:48] a=rtpmap:110 speex/8000
- [Jan 16 14:41:48] a=rtpmap:117 speex/16000
- [Jan 16 14:41:48] a=rtpmap:119 speex/32000
- [Jan 16 14:41:48] a=rtpmap:97 iLBC/8000
- [Jan 16 14:41:48] a=fmtp:97 mode=0
- [Jan 16 14:41:48] a=rtpmap:102 G7221/16000
- [Jan 16 14:41:48] a=fmtp:102 bitrate=32000
- [Jan 16 14:41:48] a=rtpmap:115 G7221/32000
- [Jan 16 14:41:48] a=fmtp:115 bitrate=48000
- [Jan 16 14:41:48] a=rtpmap:116 G719/48000
- [Jan 16 14:41:48] a=fmtp:116 bitrate=64000
- [Jan 16 14:41:48] a=rtpmap:107 opus/48000/2
- [Jan 16 14:41:48] a=rtpmap:101 telephone-event/8000
- [Jan 16 14:41:48] a=fmtp:101 0-16
- [Jan 16 14:41:48] a=maxptime:20
- [Jan 16 14:41:48] a=sendrecv
- [Jan 16 14:41:48]
- [Jan 16 14:41:48] <------------>
- [Jan 16 14:41:49] Retransmitting #1 (no NAT) to 192.168.0.52:5060:
- [Jan 16 14:41:49] NOTIFY sip:[email protected]:5060 SIP/2.0
- [Jan 16 14:41:49] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK3f8e0101;rport
- [Jan 16 14:41:49] Max-Forwards: 70
- [Jan 16 14:41:49] From: <sip:[email protected]>;tag=as45326187
- [Jan 16 14:41:49] To: <sip:[email protected]>;tag=fv4lqpihz2
- [Jan 16 14:41:49] Contact: <sip:[email protected]:5060>
- [Jan 16 14:41:49] Call-ID: 54b910bee022-dgmsn8yhawk1
- [Jan 16 14:41:49] CSeq: 117 NOTIFY
- [Jan 16 14:41:49] User-Agent: Asterisk PBX 13.1.0
- [Jan 16 14:41:49] Subscription-State: active
- [Jan 16 14:41:49] Event: dialog
- [Jan 16 14:41:49] Content-Type: application/dialog-info+xml
- [Jan 16 14:41:49] Content-Length: 201
- [Jan 16 14:41:49]
- [Jan 16 14:41:49] <?xml version="1.0"?>
- [Jan 16 14:41:49] <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="15" state="full" entity="sip:[email protected]">
- [Jan 16 14:41:49] <dialog id="91">
- [Jan 16 14:41:49] <state>confirmed</state>
- [Jan 16 14:41:49] </dialog>
- [Jan 16 14:41:49] </dialog-info>
- [Jan 16 14:41:49]
- [Jan 16 14:41:49] ---
- [Jan 16 14:41:49]
- [Jan 16 14:41:49] <--- SIP read from UDP:192.168.0.52:5060 --->
- [Jan 16 14:41:49] SIP/2.0 200 Ok
- [Jan 16 14:41:49] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK3f8e0101;rport=5060
- [Jan 16 14:41:49] From: <sip:[email protected]>;tag=as45326187
- [Jan 16 14:41:49] To: <sip:[email protected]>;tag=fv4lqpihz2
- [Jan 16 14:41:49] Call-ID: 54b910bee022-dgmsn8yhawk1
- [Jan 16 14:41:49] CSeq: 117 NOTIFY
- [Jan 16 14:41:49] Content-Length: 0
- [Jan 16 14:41:49]
- [Jan 16 14:41:49] <------------->
- [Jan 16 14:41:49] --- (7 headers 0 lines) ---
- [Jan 16 14:41:49] Retransmitting #1 (no NAT) to 192.168.0.52:5060:
- [Jan 16 14:41:49] SIP/2.0 200 OK
- [Jan 16 14:41:49] Via: SIP/2.0/UDP 192.168.0.52:5060;branch=z9hG4bK-62pgfuna2gj0;received=192.168.0.52;rport=5060
- [Jan 16 14:41:49] From: "Buero" <sip:[email protected]>;tag=4dl93g4h84
- [Jan 16 14:41:49] To: <sip:[email protected]>;tag=as581dc455
- [Jan 16 14:41:49] Call-ID: 54b915195f41-tvx8stu7cwdi
- [Jan 16 14:41:49] CSeq: 2 INVITE
- [Jan 16 14:41:49] Server: Asterisk PBX 13.1.0
- [Jan 16 14:41:49] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- [Jan 16 14:41:49] Supported: replaces, timer
- [Jan 16 14:41:49] Session-Expires: 1800;refresher=uas
- [Jan 16 14:41:49] Contact: <sip:[email protected]:5060>
- [Jan 16 14:41:49] Content-Type: application/sdp
- [Jan 16 14:41:49] Require: timer
- [Jan 16 14:41:49] Content-Length: 896
- [Jan 16 14:41:49]
- [Jan 16 14:41:49] v=0
- [Jan 16 14:41:49] o=root 1979030603 1979030603 IN IP4 192.168.0.20
- [Jan 16 14:41:49] s=Asterisk PBX 13.1.0
- [Jan 16 14:41:49] c=IN IP4 192.168.0.20
- [Jan 16 14:41:49] t=0 0
- [Jan 16 14:41:49] m=audio 10700 RTP/AVP 0 8 3 99 108 18 9 4 5 10 118 7 110 117 119 97 102 115 116 107 101
- [Jan 16 14:41:49] a=rtpmap:0 PCMU/8000
- [Jan 16 14:41:49] a=rtpmap:8 PCMA/8000
- [Jan 16 14:41:49] a=rtpmap:3 GSM/8000
- [Jan 16 14:41:49] a=rtpmap:99 G726-32/8000
- [Jan 16 14:41:49] a=rtpmap:108 AAL2-G726-32/8000
- [Jan 16 14:41:49] a=rtpmap:18 G729/8000
- [Jan 16 14:41:49] a=fmtp:18 annexb=no
- [Jan 16 14:41:49] a=rtpmap:9 G722/8000
- [Jan 16 14:41:49] a=rtpmap:4 G723/8000
- [Jan 16 14:41:49] a=fmtp:4 annexa=no
- [Jan 16 14:41:49] a=rtpmap:5 DVI4/8000
- [Jan 16 14:41:49] a=rtpmap:10 L16/8000
- [Jan 16 14:41:49] a=rtpmap:118 L16/16000
- [Jan 16 14:41:49] a=rtpmap:7 LPC/8000
- [Jan 16 14:41:49] a=rtpmap:110 speex/8000
- [Jan 16 14:41:49] a=rtpmap:117 speex/16000
- [Jan 16 14:41:49] a=rtpmap:119 speex/32000
- [Jan 16 14:41:49] a=rtpmap:97 iLBC/8000
- [Jan 16 14:41:49] a=fmtp:97 mode=0
- [Jan 16 14:41:49] a=rtpmap:102 G7221/16000
- [Jan 16 14:41:49] a=fmtp:102 bitrate=32000
- [Jan 16 14:41:49] a=rtpmap:115 G7221/32000
- [Jan 16 14:41:49] a=fmtp:115 bitrate=48000
- [Jan 16 14:41:49] a=rtpmap:116 G719/48000
- [Jan 16 14:41:49] a=fmtp:116 bitrate=64000
- [Jan 16 14:41:49] a=rtpmap:107 opus/48000/2
- [Jan 16 14:41:49] a=rtpmap:101 telephone-event/8000
- [Jan 16 14:41:49] a=fmtp:101 0-16
- [Jan 16 14:41:49] a=maxptime:20
- [Jan 16 14:41:49] a=sendrecv
- [Jan 16 14:41:49]
- [Jan 16 14:41:49] ---
- [Jan 16 14:41:49]
- [Jan 16 14:41:49] <--- SIP read from UDP:192.168.0.52:5060 --->
- [Jan 16 14:41:49] ACK sip:[email protected]:5060 SIP/2.0
- [Jan 16 14:41:49] Via: SIP/2.0/UDP 192.168.0.52:5060;branch=z9hG4bK-nrkwwcisv15i;rport
- [Jan 16 14:41:49] From: "Buero" <sip:[email protected]>;tag=4dl93g4h84
- [Jan 16 14:41:49] To: <sip:[email protected]>;tag=as581dc455
- [Jan 16 14:41:49] Call-ID: 54b915195f41-tvx8stu7cwdi
- [Jan 16 14:41:49] CSeq: 2 ACK
- [Jan 16 14:41:49] Max-Forwards: 70
- [Jan 16 14:41:49] Contact: <sip:[email protected]:5060>;reg-id=1
- [Jan 16 14:41:49] Content-Length: 0
- [Jan 16 14:41:49]
- [Jan 16 14:41:49] <------------->
- [Jan 16 14:41:49] --- (9 headers 0 lines) ---
- [Jan 16 14:41:49]
- [Jan 16 14:41:49] <--- SIP read from UDP:192.168.0.52:5060 --->
- [Jan 16 14:41:49] SIP/2.0 200 Ok
- [Jan 16 14:41:49] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK3f8e0101;rport=5060
- [Jan 16 14:41:49] From: <sip:[email protected]>;tag=as45326187
- [Jan 16 14:41:49] To: <sip:[email protected]>;tag=fv4lqpihz2
- [Jan 16 14:41:49] Call-ID: 54b910bee022-dgmsn8yhawk1
- [Jan 16 14:41:49] CSeq: 117 NOTIFY
- [Jan 16 14:41:49] Content-Length: 0
- [Jan 16 14:41:49]
- [Jan 16 14:41:49] <------------->
- [Jan 16 14:41:49] --- (7 headers 0 lines) ---
- [Jan 16 14:41:49] -- <SIP/91-00000063> Playing 'vm-youhave.ulaw' (language 'en')
- [Jan 16 14:41:49] > 0x7f88480d42c0 -- Probation passed - setting RTP source address to 192.168.0.52:56520
- [Jan 16 14:41:49]
- [Jan 16 14:41:49] <--- SIP read from UDP:192.168.0.52:5060 --->
- [Jan 16 14:41:49] ACK sip:[email protected]:5060 SIP/2.0
- [Jan 16 14:41:49] Via: SIP/2.0/UDP 192.168.0.52:5060;branch=z9hG4bK-nrkwwcisv15i;rport
- [Jan 16 14:41:49] From: "Buero" <sip:[email protected]>;tag=4dl93g4h84
- [Jan 16 14:41:49] To: <sip:[email protected]>;tag=as581dc455
- [Jan 16 14:41:49] Call-ID: 54b915195f41-tvx8stu7cwdi
- [Jan 16 14:41:49] CSeq: 2 ACK
- [Jan 16 14:41:49] Max-Forwards: 70
- [Jan 16 14:41:49] Contact: <sip:[email protected]:5060>;reg-id=1
- [Jan 16 14:41:49] Content-Length: 0
- [Jan 16 14:41:49]
- [Jan 16 14:41:49] <------------->
- [Jan 16 14:41:49] --- (9 headers 0 lines) ---
- [Jan 16 14:41:49] Retransmitting #6 (no NAT) to 173.208.168.34:5074:
- [Jan 16 14:41:49] SIP/2.0 401 Unauthorized
- [Jan 16 14:41:49] Via: SIP/2.0/UDP 173.208.168.34:5074;branch=z9hG4bK-94b11f985714486caa3d93367bc54d21;received=173.208.168.34;rport=5074
- [Jan 16 14:41:49] From: 5001<sip:[email protected]>;tag=5d52d395
- [Jan 16 14:41:49] To: 9009441904891104<sip:[email protected]>;tag=as7f50d10b
- [Jan 16 14:41:49] Call-ID: 94b11f985714486caa3d93367bc54d21
- [Jan 16 14:41:49] CSeq: 1 INVITE
- [Jan 16 14:41:49] Server: Asterisk PBX 13.1.0
- [Jan 16 14:41:49] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- [Jan 16 14:41:49] Supported: replaces, timer
- [Jan 16 14:41:49] WWW-Authenticate: Digest algorithm=MD5, realm="kaworu.kunbox.net", nonce="7770a9b0"
- [Jan 16 14:41:49] Content-Length: 0
- [Jan 16 14:41:49]
- [Jan 16 14:41:49]
- [Jan 16 14:41:49] ---
- [Jan 16 14:41:50] -- <SIP/91-00000063> Playing 'digits/1.ulaw' (language 'en')
- [Jan 16 14:41:50] -- <SIP/91-00000063> Playing 'vm-INBOX.ulaw' (language 'en')
- [Jan 16 14:41:51] -- <SIP/91-00000063> Playing 'vm-message.ulaw' (language 'en')
- [Jan 16 14:41:52] Reliably Transmitting (no NAT) to 192.168.0.100:4224:
- [Jan 16 14:41:52] OPTIONS sip:[email protected]:4224;rinstance=7ac320eaa202f4fe SIP/2.0
- [Jan 16 14:41:52] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK5e1cd622
- [Jan 16 14:41:52] Max-Forwards: 70
- [Jan 16 14:41:52] From: "asterisk" <sip:[email protected]>;tag=as647df5b2
- [Jan 16 14:41:52] To: <sip:[email protected]:4224;rinstance=7ac320eaa202f4fe>
- [Jan 16 14:41:52] Contact: <sip:[email protected]:5060>
- [Jan 16 14:41:52] Call-ID: [email protected]:5060
- [Jan 16 14:41:52] CSeq: 102 OPTIONS
- [Jan 16 14:41:52] User-Agent: Asterisk PBX 13.1.0
- [Jan 16 14:41:52] Date: Fri, 16 Jan 2015 13:41:52 GMT
- [Jan 16 14:41:52] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- [Jan 16 14:41:52] Supported: replaces, timer
- [Jan 16 14:41:52] Content-Length: 0
- [Jan 16 14:41:52]
- [Jan 16 14:41:52]
- [Jan 16 14:41:52] ---
- [Jan 16 14:41:52]
- [Jan 16 14:41:52] <--- SIP read from UDP:192.168.0.100:4224 --->
- [Jan 16 14:41:52] SIP/2.0 200 OK
- [Jan 16 14:41:52] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK5e1cd622
- [Jan 16 14:41:52] Contact: <sip:192.168.0.100:4224>
- [Jan 16 14:41:52] To: <sip:[email protected]:4224;rinstance=7ac320eaa202f4fe>;tag=61884252
- [Jan 16 14:41:52] From: "asterisk"<sip:[email protected]>;tag=as647df5b2
- [Jan 16 14:41:52] Call-ID: [email protected]:5060
- [Jan 16 14:41:52] CSeq: 102 OPTIONS
- [Jan 16 14:41:52] Accept: application/sdp
- [Jan 16 14:41:52] Accept-Language: en
- [Jan 16 14:41:52] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
- [Jan 16 14:41:52] Supported: replaces, eventlist
- [Jan 16 14:41:52] User-Agent: X-Lite release 4.7.1 stamp 74250
- [Jan 16 14:41:52] Content-Length: 0
- [Jan 16 14:41:52]
- [Jan 16 14:41:52] <------------->
- [Jan 16 14:41:52] --- (13 headers 0 lines) ---
- [Jan 16 14:41:52] Really destroying SIP dialog '[email protected]:5060' Method: OPTIONS
- [Jan 16 14:41:52] -- <SIP/91-00000063> Playing 'vm-onefor.ulaw' (language 'en')
- [Jan 16 14:41:53] -- <SIP/91-00000063> Playing 'vm-INBOX.ulaw' (language 'en')
- [Jan 16 14:41:53] Retransmitting #7 (no NAT) to 173.208.168.34:5074:
- [Jan 16 14:41:53] SIP/2.0 401 Unauthorized
- [Jan 16 14:41:53] Via: SIP/2.0/UDP 173.208.168.34:5074;branch=z9hG4bK-94b11f985714486caa3d93367bc54d21;received=173.208.168.34;rport=5074
- [Jan 16 14:41:53] From: 5001<sip:[email protected]>;tag=5d52d395
- [Jan 16 14:41:53] To: 9009441904891104<sip:[email protected]>;tag=as7f50d10b
- [Jan 16 14:41:53] Call-ID: 94b11f985714486caa3d93367bc54d21
- [Jan 16 14:41:53] CSeq: 1 INVITE
- [Jan 16 14:41:53] Server: Asterisk PBX 13.1.0
- [Jan 16 14:41:53] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- [Jan 16 14:41:53] Supported: replaces, timer
- [Jan 16 14:41:53] WWW-Authenticate: Digest algorithm=MD5, realm="kaworu.kunbox.net", nonce="7770a9b0"
- [Jan 16 14:41:53] Content-Length: 0
- [Jan 16 14:41:53]
- [Jan 16 14:41:53]
- [Jan 16 14:41:53] ---
- [Jan 16 14:41:54] -- <SIP/91-00000063> Playing 'vm-messages.ulaw' (language 'en')
- [Jan 16 14:41:55] -- <SIP/91-00000063> Playing 'vm-opts.ulaw' (language 'en')
- [Jan 16 14:41:56]
- [Jan 16 14:41:56] <--- SIP read from UDP:192.168.0.100:4224 --->
- [Jan 16 14:41:56]
- [Jan 16 14:41:56]
- [Jan 16 14:41:56] <------------->
- [Jan 16 14:41:56] -- <SIP/91-00000063> Playing 'vm-first.ulaw' (language 'en')
- [Jan 16 14:41:57] == Parsing '/var/spool/asterisk/voicemail/default/800/INBOX/msg0000.txt': Found
- [Jan 16 14:41:57] -- <SIP/91-00000063> Playing 'vm-message.ulaw' (language 'en')
- [Jan 16 14:41:57] Retransmitting #8 (no NAT) to 173.208.168.34:5074:
- [Jan 16 14:41:57] SIP/2.0 401 Unauthorized
- [Jan 16 14:41:57] Via: SIP/2.0/UDP 173.208.168.34:5074;branch=z9hG4bK-94b11f985714486caa3d93367bc54d21;received=173.208.168.34;rport=5074
- [Jan 16 14:41:57] From: 5001<sip:[email protected]>;tag=5d52d395
- [Jan 16 14:41:57] To: 9009441904891104<sip:[email protected]>;tag=as7f50d10b
- [Jan 16 14:41:57] Call-ID: 94b11f985714486caa3d93367bc54d21
- [Jan 16 14:41:57] CSeq: 1 INVITE
- [Jan 16 14:41:57] Server: Asterisk PBX 13.1.0
- [Jan 16 14:41:57] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
- [Jan 16 14:41:57] Supported: replaces, timer
- [Jan 16 14:41:57] WWW-Authenticate: Digest algorithm=MD5, realm="kaworu.kunbox.net", nonce="7770a9b0"
- [Jan 16 14:41:57] Content-Length: 0
- [Jan 16 14:41:57]
- [Jan 16 14:41:57]
- [Jan 16 14:41:57] ---
- [Jan 16 14:41:58] Really destroying SIP dialog 'v9tDi~xRt0' Method: REGISTER
- [Jan 16 14:41:58] == Spawn extension (internal, 98, 1) exited non-zero on 'SIP/91-00000063'
- [Jan 16 14:41:58] Scheduling destruction of SIP dialog '54b915195f41-tvx8stu7cwdi' in 6400 ms (Method: ACK)
- [Jan 16 14:41:58] set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
- [Jan 16 14:41:58] set_destination: set destination to 192.168.0.51:5060
- [Jan 16 14:41:58] Reliably Transmitting (no NAT) to 192.168.0.51:5060:
- [Jan 16 14:41:58] NOTIFY sip:[email protected]:5060 SIP/2.0
- [Jan 16 14:41:58] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK7c31750b;rport
- [Jan 16 14:41:58] Max-Forwards: 70
- [Jan 16 14:41:58] From: <sip:[email protected]>;tag=as3be3a100
- [Jan 16 14:41:58] To: <sip:[email protected]>;tag=yqam73tqu3
- [Jan 16 14:41:58] Contact: <sip:[email protected]:5060>
- [Jan 16 14:41:58] Call-ID: 386d43a9e00b-xh9vubfgjat8
- [Jan 16 14:41:58] CSeq: 181 NOTIFY
- [Jan 16 14:41:58] User-Agent: Asterisk PBX 13.1.0
- [Jan 16 14:41:58] Subscription-State: active
- [Jan 16 14:41:58] Event: dialog
- [Jan 16 14:41:58] Content-Type: application/dialog-info+xml
- [Jan 16 14:41:58] Content-Length: 202
- [Jan 16 14:41:58]
- [Jan 16 14:41:58] <?xml version="1.0"?>
- [Jan 16 14:41:58] <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="79" state="full" entity="sip:[email protected]">
- [Jan 16 14:41:58] <dialog id="91">
- [Jan 16 14:41:58] <state>terminated</state>
- [Jan 16 14:41:58] </dialog>
- [Jan 16 14:41:58] </dialog-info>
- [Jan 16 14:41:58]
- [Jan 16 14:41:58] ---
- [Jan 16 14:41:58] == Extension Changed 91[SoftPhone] new state Idle for Notify User 76
- [Jan 16 14:41:58] set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
- [Jan 16 14:41:58] set_destination: set destination to 192.168.0.52:5060
- [Jan 16 14:41:58] Reliably Transmitting (no NAT) to 192.168.0.52:5060:
- [Jan 16 14:41:58] NOTIFY sip:[email protected]:5060 SIP/2.0
- [Jan 16 14:41:58] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK740271c5;rport
- [Jan 16 14:41:58] Max-Forwards: 70
- [Jan 16 14:41:58] From: <sip:[email protected]>;tag=as45326187
- [Jan 16 14:41:58] To: <sip:[email protected]>;tag=fv4lqpihz2
- [Jan 16 14:41:58] Contact: <sip:[email protected]:5060>
- [Jan 16 14:41:58] Call-ID: 54b910bee022-dgmsn8yhawk1
- [Jan 16 14:41:58] CSeq: 118 NOTIFY
- [Jan 16 14:41:58] User-Agent: Asterisk PBX 13.1.0
- [Jan 16 14:41:58] Subscription-State: active
- [Jan 16 14:41:58] Event: dialog
- [Jan 16 14:41:58] Content-Type: application/dialog-info+xml
- [Jan 16 14:41:58] Content-Length: 202
- [Jan 16 14:41:58]
- [Jan 16 14:41:58] <?xml version="1.0"?>
- [Jan 16 14:41:58] <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="16" state="full" entity="sip:[email protected]">
- [Jan 16 14:41:58] <dialog id="91">
- [Jan 16 14:41:58] <state>terminated</state>
- [Jan 16 14:41:58] </dialog>
- [Jan 16 14:41:58] </dialog-info>
- [Jan 16 14:41:58]
- [Jan 16 14:41:58] ---
- [Jan 16 14:41:58] == Extension Changed 91[SoftPhone] new state Idle for Notify User 91
- [Jan 16 14:41:58] set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
- [Jan 16 14:41:58] set_destination: set destination to 192.168.0.52:5060
- [Jan 16 14:41:58] Reliably Transmitting (no NAT) to 192.168.0.52:5060:
- [Jan 16 14:41:58] BYE sip:[email protected]:5060 SIP/2.0
- [Jan 16 14:41:58] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK615b412b;rport
- [Jan 16 14:41:58] Max-Forwards: 70
- [Jan 16 14:41:58] From: <sip:[email protected]>;tag=as581dc455
- [Jan 16 14:41:58] To: "Buero" <sip:[email protected]>;tag=4dl93g4h84
- [Jan 16 14:41:58] Call-ID: 54b915195f41-tvx8stu7cwdi
- [Jan 16 14:41:58] CSeq: 102 BYE
- [Jan 16 14:41:58] User-Agent: Asterisk PBX 13.1.0
- [Jan 16 14:41:58] Proxy-Authorization: Digest username="91", realm="kaworu.kunbox.net", algorithm=MD5, uri="sip:192.168.0.20", nonce="75387cce", response="fab253b48214658abde24db429c88d78"
- [Jan 16 14:41:58] X-Asterisk-HangupCause: Unknown
- [Jan 16 14:41:58] X-Asterisk-HangupCauseCode: 0
- [Jan 16 14:41:58] Content-Length: 0
- [Jan 16 14:41:58]
- [Jan 16 14:41:58]
- [Jan 16 14:41:58] ---
- [Jan 16 14:41:58]
- [Jan 16 14:41:58] <--- SIP read from UDP:192.168.0.51:5060 --->
- [Jan 16 14:41:58] SIP/2.0 200 Ok
- [Jan 16 14:41:58] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK7c31750b;rport=5060
- [Jan 16 14:41:58] From: <sip:[email protected]>;tag=as3be3a100
- [Jan 16 14:41:58] To: <sip:[email protected]>;tag=yqam73tqu3
- [Jan 16 14:41:58] Call-ID: 386d43a9e00b-xh9vubfgjat8
- [Jan 16 14:41:58] CSeq: 181 NOTIFY
- [Jan 16 14:41:58] Content-Length: 0
- [Jan 16 14:41:58]
- [Jan 16 14:41:58] <------------->
- [Jan 16 14:41:58] --- (7 headers 0 lines) ---
- [Jan 16 14:41:58]
- [Jan 16 14:41:58] <--- SIP read from UDP:192.168.0.52:5060 --->
- [Jan 16 14:41:58] SIP/2.0 200 Ok
- [Jan 16 14:41:58] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK740271c5;rport=5060
- [Jan 16 14:41:58] From: <sip:[email protected]>;tag=as45326187
- [Jan 16 14:41:58] To: <sip:[email protected]>;tag=fv4lqpihz2
- [Jan 16 14:41:58] Call-ID: 54b910bee022-dgmsn8yhawk1
- [Jan 16 14:41:58] CSeq: 118 NOTIFY
- [Jan 16 14:41:58] Content-Length: 0
- [Jan 16 14:41:58]
- [Jan 16 14:41:58] <------------->
- [Jan 16 14:41:58] --- (7 headers 0 lines) ---
- [Jan 16 14:41:58]
- [Jan 16 14:41:58] <--- SIP read from UDP:192.168.0.52:5060 --->
- [Jan 16 14:41:58] SIP/2.0 200 OK
- [Jan 16 14:41:58] Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK615b412b;rport=5060
- [Jan 16 14:41:58] From: <sip:[email protected]>;tag=as581dc455
- [Jan 16 14:41:58] To: "Buero" <sip:[email protected]>;tag=4dl93g4h84
- [Jan 16 14:41:58] Call-ID: 54b915195f41-tvx8stu7cwdi
- [Jan 16 14:41:58] CSeq: 102 BYE
- [Jan 16 14:41:58] Contact: <sip:[email protected]:5060>;reg-id=1
- [Jan 16 14:41:58] User-Agent: snom360/8.7.3.25
- [Jan 16 14:41:58] RTP-RxStat: Total_Rx_Pkts=455,Rx_Pkts=0,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0
- [Jan 16 14:41:58] RTP-TxStat: Total_Tx_Pkts=454,Tx_Pkts=454,Remote_Tx_Pkts=253
- [Jan 16 14:41:58] Content-Length: 0
- [Jan 16 14:41:58]
- [Jan 16 14:41:58] <------------->
Advertisement
Add Comment
Please, Sign In to add comment
Advertisement