rendered paste bodyasteriskRecorder*CLI>
<--- SIP read from 192.168.1.104:5060 --->
REGISTER sip:192.168.1.27 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.104:5060;rport;branch=z9hG4bK7C8A4951775D4C2799C9EEFBB72AEDBA
From: Roderick <sip:9910001@192.168.1.27>;tag=1848079261
To: Roderick <sip:9910001@192.168.1.27>
Contact: "Roderick" <sip:9910001@192.168.1.104:5060>
Call-ID: 100FD5584507403DA28BC927854DD24F@192.168.1.27
CSeq: 27961 REGISTER
Expires: 60
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.1.104 : 5060 (NAT)
<--- Transmitting (no NAT) to 192.168.1.104:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK7C8A4951775D4C2799C9EEFBB72AEDBA;received=192.168.1.104;rport=5060
From: Roderick <sip:9910001@192.168.1.27>;tag=1848079261
To: Roderick <sip:9910001@192.168.1.27>
Call-ID: 100FD5584507403DA28BC927854DD24F@192.168.1.27
CSeq: 27961 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:9910001@192.168.1.27:5065>
Content-Length: 0
<------------>
asteriskRecorder*CLI>
<--- Transmitting (no NAT) to 192.168.1.104:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK7C8A4951775D4C2799C9EEFBB72AEDBA;received=192.168.1.104;rport=5060
From: Roderick <sip:9910001@192.168.1.27>;tag=1848079261
To: Roderick <sip:9910001@192.168.1.27>;tag=as5ff8a78b
Call-ID: 100FD5584507403DA28BC927854DD24F@192.168.1.27
CSeq: 27961 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 60
Contact: <sip:9910001@192.168.1.104:5060>;expires=60
Date: Tue, 26 Jun 2007 19:37:06 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '100FD5584507403DA28BC927854DD24F@192.168.1.27' in 32000 ms (Method: REGISTER)
asteriskRecorder*CLI>
<--- SIP read from 192.168.1.100:5060 --->
INVITE sip:123456@192.168.1.27:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK7614d594a11d3d7d
From: <sip:279@192.168.1.27:5065>;tag=ddfe4836d6e8ad6e
To: <sip:123456@192.168.1.27:5065>
Contact: <sip:279@192.168.1.100>
Supported: replaces
Call-ID: 5367e8b3e8a75b26@192.168.1.100
CSeq: 6184 INVITE
User-Agent: Grandstream HT286 1.0.7.19
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 156
v=0
o=279 8000 8000 IN IP4 192.168.1.100
s=SIP Call
c=IN IP4 192.168.1.100
t=0 0
m=audio 8002 RTP/AVP 0
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:40
<------------->
--- (13 headers 9 lines) ---
Sending to 192.168.1.100 : 5060 (no NAT)
Using INVITE request as basis request - 5367e8b3e8a75b26@192.168.1.100
Found peer '279'
Found RTP audio format 0
Peer audio RTP is at port 192.168.1.100:8002
Found description format PCMU for ID 0
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.100:8002
Looking for 123456 in default (domain 192.168.1.27)
list_route: hop: <sip:279@192.168.1.100>
<--- Transmitting (no NAT) to 192.168.1.100:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK7614d594a11d3d7d;received=192.168.1.100
From: <sip:279@192.168.1.27:5065>;tag=ddfe4836d6e8ad6e
To: <sip:123456@192.168.1.27:5065>
Call-ID: 5367e8b3e8a75b26@192.168.1.100
CSeq: 6184 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:123456@192.168.1.27:5065>
Content-Length: 0
<------------>
-- Executing [123456@default:1] Set("SIP/279-081dcad0", "MONITOR_EXEC=/etc/asterisk/soxmixwav.sh") in new stack
-- Executing [123456@default:2] Playback("SIP/279-081dcad0", "sera_grabada") in new stack
Audio is at 192.168.1.27 port 10652
Adding codec 0x4 (ulaw) to SDP
<--- Reliably Transmitting (no NAT) to 192.168.1.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK7614d594a11d3d7d;received=192.168.1.100
From: <sip:279@192.168.1.27:5065>;tag=ddfe4836d6e8ad6e
To: <sip:123456@192.168.1.27:5065>;tag=as5ea476c4
Call-ID: 5367e8b3e8a75b26@192.168.1.100
CSeq: 6184 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:123456@192.168.1.27:5065>
Content-Type: application/sdp
Content-Length: 182
v=0
o=root 6675 6675 IN IP4 192.168.1.27
s=session
c=IN IP4 192.168.1.27
t=0 0
m=audio 10652 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
-- <SIP/279-081dcad0> Playing 'sera_grabada' (language 'en')
asteriskRecorder*CLI>
<--- SIP read from 192.168.1.100:5060 --->
ACK sip:123456@192.168.1.27:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK5baea0d15ebefa6d
From: <sip:279@192.168.1.27:5065>;tag=ddfe4836d6e8ad6e
To: <sip:123456@192.168.1.27:5065>;tag=as5ea476c4
Contact: <sip:279@192.168.1.100>
Call-ID: 5367e8b3e8a75b26@192.168.1.100
CSeq: 6184 ACK
User-Agent: Grandstream HT286 1.0.7.19
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
-- Executing [123456@default:3] Answer("SIP/279-081dcad0", "") in new stack
-- Executing [123456@default:4] Monitor("SIP/279-081dcad0", "wav|123456") in new stack
-- Executing [123456@default:5] Dial("SIP/279-081dcad0", "SIP/123456") in new stack
Audio is at 192.168.1.27 port 16866
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (no NAT) to 192.168.1.27:5060:
INVITE sip:123456@192.168.1.27:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK30e306a3;rport
From: "279" <sip:279@192.168.1.27:5065>;tag=as3b74adbb
To: <sip:123456@192.168.1.27:5060>
Contact: <sip:279@192.168.1.27:5065>
Call-ID: 1a50079e121d5a025a2a25124067b0de@192.168.1.27
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 26 Jun 2007 19:37:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 229
v=0
o=root 6675 6675 IN IP4 192.168.1.27
s=session
c=IN IP4 192.168.1.27
t=0 0
m=audio 16866 RTP/AVP 0 3 8
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 123456
asteriskRecorder*CLI>
<--- SIP read from 192.168.1.27:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK30e306a3;rport
From: "279" <sip:279@192.168.1.27:5065>;tag=as3b74adbb
To: <sip:123456@192.168.1.27:5060>;tag=1984039473
Contact: <sip:123456@192.168.1.27:5060>
Call-ID: 1a50079e121d5a025a2a25124067b0de@192.168.1.27
CSeq: 102 INVITE
Server: X-Lite release 1105d
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from 192.168.1.27:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK30e306a3;rport
From: "279" <sip:279@192.168.1.27:5065>;tag=as3b74adbb
To: <sip:123456@192.168.1.27:5060>;tag=1984039473
Contact: <sip:123456@192.168.1.27:5060>
Call-ID: 1a50079e121d5a025a2a25124067b0de@192.168.1.27
CSeq: 102 INVITE
Server: X-Lite release 1105d
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
-- SIP/123456-081d7480 is ringing
set_destination: Parsing <sip:279@192.168.1.100> for address/port to send to
set_destination: set destination to 192.168.1.100, port 5060
Audio is at 192.168.1.27 port 10652
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 192.168.1.100:5060:
INVITE sip:279@192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK7f7dc7bc;rport
From: <sip:123456@192.168.1.27:5065>;tag=as5ea476c4
To: <sip:279@192.168.1.27:5065>;tag=ddfe4836d6e8ad6e
Contact: <sip:123456@192.168.1.27:5065>
Call-ID: 5367e8b3e8a75b26@192.168.1.100
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 182
v=0
o=root 6675 6676 IN IP4 192.168.1.27
s=session
c=IN IP4 192.168.1.27
t=0 0
m=audio 10652 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
<--- SIP read from 192.168.1.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK7f7dc7bc;rport
From: <sip:123456@192.168.1.27:5065>;tag=as5ea476c4
To: <sip:279@192.168.1.27:5065>;tag=ddfe4836d6e8ad6e
Call-ID: 5367e8b3e8a75b26@192.168.1.100
CSeq: 102 INVITE
User-Agent: Grandstream HT286 1.0.7.19
Contact: <sip:279@192.168.1.100>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Supported: replaces
Content-Length: 156
v=0
o=279 8000 8001 IN IP4 192.168.1.100
s=SIP Call
c=IN IP4 192.168.1.100
t=0 0
m=audio 8002 RTP/AVP 0
=sendrecvecorder*CLI>
a=rtpmap:0 PCMU/8000
a=ptime:20
<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 0
Peer audio RTP is at port 192.168.1.100:8002
Found description format PCMU for ID 0
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.100:8002
list_route: hop: <sip:279@192.168.1.100>
set_destination: Parsing <sip:279@192.168.1.100> for address/port to send to
set_destination: set destination to 192.168.1.100, port 5060
Transmitting (no NAT) to 192.168.1.100:5060:
ACK sip:279@192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK580b5810;rport
From: <sip:123456@192.168.1.27:5065>;tag=as5ea476c4
To: <sip:279@192.168.1.27:5065>;tag=ddfe4836d6e8ad6e
Contact: <sip:123456@192.168.1.27:5065>
Call-ID: 5367e8b3e8a75b26@192.168.1.100
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
asteriskRecorder*CLI>
<--- SIP read from 192.168.1.27:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK30e306a3;rport
From: "279" <sip:279@192.168.1.27:5065>;tag=as3b74adbb
To: <sip:123456@192.168.1.27:5060>;tag=1984039473
Contact: <sip:123456@192.168.1.27:5060>
Call-ID: 1a50079e121d5a025a2a25124067b0de@192.168.1.27
CSeq: 102 INVITE
Content-Type: application/sdp
Server: X-Lite release 1105d
Content-Length: 311
v=0
o=123456 1770631733 1770638493 IN IP4 192.168.1.27
s=X-Lite
c=IN IP4 192.168.1.27
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (10 headers 14 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.27:8000
Found description format pcmu for ID 0
Found description format pcma for ID 8
Found description format gsm for ID 3
Found description format iLBC for ID 98
Found description format speex for ID 97
Found description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.27:8000
list_route: hop: <sip:123456@192.168.1.27:5060>
set_destination: Parsing <sip:123456@192.168.1.27:5060> for address/port to send to
set_destination: set destination to 192.168.1.27, port 5060
Transmitting (no NAT) to 192.168.1.27:5060:
ACK sip:123456@192.168.1.27:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK689c9fce;rport
From: "279" <sip:279@192.168.1.27:5065>;tag=as3b74adbb
To: <sip:123456@192.168.1.27:5060>;tag=1984039473
Contact: <sip:279@192.168.1.27:5065>
Call-ID: 1a50079e121d5a025a2a25124067b0de@192.168.1.27
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/123456-081d7480 answered SIP/279-081dcad0
set_destination: Parsing <sip:279@192.168.1.100> for address/port to send to
set_destination: set destination to 192.168.1.100, port 5060
Audio is at 192.168.1.27 port 10652
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 192.168.1.100:5060:
INVITE sip:279@192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK24ebc30f;rport
From: <sip:123456@192.168.1.27:5065>;tag=as5ea476c4
To: <sip:279@192.168.1.27:5065>;tag=ddfe4836d6e8ad6e
Contact: <sip:123456@192.168.1.27:5065>
Call-ID: 5367e8b3e8a75b26@192.168.1.100
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 181
v=0
o=root 6675 6677 IN IP4 192.168.1.27
s=session
c=IN IP4 192.168.1.27
t=0 0
m=audio 8000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
asteriskRecorder*CLI>
<--- SIP read from 192.168.1.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK24ebc30f;rport
From: <sip:123456@192.168.1.27:5065>;tag=as5ea476c4
To: <sip:279@192.168.1.27:5065>;tag=ddfe4836d6e8ad6e
Call-ID: 5367e8b3e8a75b26@192.168.1.100
CSeq: 103 INVITE
User-Agent: Grandstream HT286 1.0.7.19
Contact: <sip:279@192.168.1.100>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Supported: replaces
Content-Length: 156
v=0
o=279 8000 8002 IN IP4 192.168.1.100
s=SIP Call
c=IN IP4 192.168.1.100
t=0 0
m=audio 8002 RTP/AVP 0
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 0
Peer audio RTP is at port 192.168.1.100:8002
Found description format PCMU for ID 0
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.100:8002
set_destination: Parsing <sip:279@192.168.1.100> for address/port to send to
set_destination: set destination to 192.168.1.100, port 5060
Transmitting (no NAT) to 192.168.1.100:5060:
ACK sip:279@192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK76774c8a;rport
From: <sip:123456@192.168.1.27:5065>;tag=as5ea476c4
To: <sip:279@192.168.1.27:5065>;tag=ddfe4836d6e8ad6e
Contact: <sip:123456@192.168.1.27:5065>
Call-ID: 5367e8b3e8a75b26@192.168.1.100
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Really destroying SIP dialog '306fcaeb1fe9e2e3@192.168.1.100' Method: REGISTER
Really destroying SIP dialog '1A58AF22A0D97CE7EDD9112B17A83AF3@192.168.1.27' Method: REGISTER
asteriskRecorder*CLI>
<--- SIP read from 192.168.1.100:5060 --->
REGISTER sip:192.168.1.27:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bKbd03fda823d73b54
From: <sip:279@192.168.1.27:5065>;tag=7ac086505cddff9f
To: <sip:279@192.168.1.27:5065>
Contact: <sip:279@192.168.1.100>
Supported: replaces
Call-ID: 306fcaeb1fe9e2e3@192.168.1.100
CSeq: 241 REGISTER
Expires: 60
User-Agent: Grandstream HT286 1.0.7.19
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
<------------->r*CLI>
--- (13 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.1.100 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.1.100:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bKbd03fda823d73b54;received=192.168.1.100
From: <sip:279@192.168.1.27:5065>;tag=7ac086505cddff9f
To: <sip:279@192.168.1.27:5065>
Call-ID: 306fcaeb1fe9e2e3@192.168.1.100
CSeq: 241 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:279@192.168.1.27:5065>
Content-Length: 0
<------------>
asteriskRecorder*CLI>
<--- Transmitting (no NAT) to 192.168.1.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bKbd03fda823d73b54;received=192.168.1.100
From: <sip:279@192.168.1.27:5065>;tag=7ac086505cddff9f
To: <sip:279@192.168.1.27:5065>;tag=as343ae754
Call-ID: 306fcaeb1fe9e2e3@192.168.1.100
CSeq: 241 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 60
Contact: <sip:279@192.168.1.100>;expires=60
Date: Tue, 26 Jun 2007 19:37:38 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '306fcaeb1fe9e2e3@192.168.1.100' in 32000 ms (Method: REGISTER)
Really destroying SIP dialog '100FD5584507403DA28BC927854DD24F@192.168.1.27' Method: REGISTER
asteriskRecorder*CLI>
<--- SIP read from 192.168.1.100:5060 --->
BYE sip:123456@192.168.1.27:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK604d7c9453a2dab4
From: <sip:279@192.168.1.27:5065>;tag=ddfe4836d6e8ad6e
To: <sip:123456@192.168.1.27:5065>;tag=as5ea476c4
Supported: replaces
Call-ID: 5367e8b3e8a75b26@192.168.1.100
CSeq: 6185 BYE
User-Agent: Grandstream HT286 1.0.7.19
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.1.100 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.1.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK604d7c9453a2dab4;received=192.168.1.100
From: <sip:279@192.168.1.27:5065>;tag=ddfe4836d6e8ad6e
To: <sip:123456@192.168.1.27:5065>;tag=as5ea476c4
Call-ID: 5367e8b3e8a75b26@192.168.1.100
CSeq: 6185 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:123456@192.168.1.27:5065>
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1a50079e121d5a025a2a25124067b0de@192.168.1.27' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:123456@192.168.1.27:5060> for address/port to send to
set_destination: set destination to 192.168.1.27, port 5060
Reliably Transmitting (no NAT) to 192.168.1.27:5060:
BYE sip:123456@192.168.1.27:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK705351c1;rport
From: "279" <sip:279@192.168.1.27:5065>;tag=as3b74adbb
To: <sip:123456@192.168.1.27:5060>;tag=1984039473
Call-ID: 1a50079e121d5a025a2a25124067b0de@192.168.1.27
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
== Spawn extension (default, 123456, 5) exited non-zero on 'SIP/279-081dcad0'
asteriskRecorder*CLI>
<--- SIP read from 192.168.1.27:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.1.27:5065;branch=z9hG4bK705351c1;rport
From: "279" <sip:279@192.168.1.27:5065>;tag=as3b74adbb
To: <sip:123456@192.168.1.27:5060>;tag=1984039473
Contact: <sip:123456@192.168.1.27:5060>
Call-ID: 1a50079e121d5a025a2a25124067b0de@192.168.1.27
CSeq: 103 BYE
Server: X-Lite release 1105d
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '1a50079e121d5a025a2a25124067b0de@192.168.1.27' Method: INVITE
Really destroying SIP dialog '5367e8b3e8a75b26@192.168.1.100' Method: BYE
asteriskRecorder*CLI>
<--- SIP read from 192.168.1.27:5060 --->
REGISTER sip:192.168.1.27 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.27:5060;rport;branch=z9hG4bK1889F1F1070BEF509A4D33B19C5FE963
From: 1234566 <sip:123456@192.168.1.27>;tag=1358852701
To: 1234566 <sip:123456@192.168.1.27>
Contact: "1234566" <sip:123456@192.168.1.27:5060>
Call-ID: 1A58AF22A0D97CE7EDD9112B17A83AF3@192.168.1.27
CSeq: 56613 REGISTER
Expires: 60
Max-Forwards: 70
User-Agent: X-Lite release 1105d
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.1.27 : 5060 (NAT)
<--- Transmitting (no NAT) to 192.168.1.27:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK1889F1F1070BEF509A4D33B19C5FE963;received=192.168.1.27;rport=5060
From: 1234566 <sip:123456@192.168.1.27>;tag=1358852701
To: 1234566 <sip:123456@192.168.1.27>
Call-ID: 1A58AF22A0D97CE7EDD9112B17A83AF3@192.168.1.27
CSeq: 56613 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:123456@192.168.1.27:5065>
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 192.168.1.27:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.27:5060;branch=z9hG4bK1889F1F1070BEF509A4D33B19C5FE963;received=192.168.1.27;rport=5060
From: 1234566 <sip:123456@192.168.1.27>;tag=1358852701
To: 1234566 <sip:123456@192.168.1.27>;tag=as3ecdfa30
Call-ID: 1A58AF22A0D97CE7EDD9112B17A83AF3@192.168.1.27
CSeq: 56613 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 60
Contact: <sip:123456@192.168.1.27:5060>;expires=60
Date: Tue, 26 Jun 2007 19:37:59 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1A58AF22A0D97CE7EDD9112B17A83AF3@192.168.1.27' in 32000 ms (Method: REGISTER)
asteriskRecorder*CLI>
<--- SIP read from 192.168.1.104:5060 --->
REGISTER sip:192.168.1.27 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.104:5060;rport;branch=z9hG4bK7FCA4317695A49B8ACB8F480819C5D02
From: Roderick <sip:9910001@192.168.1.27>;tag=1848079261
To: Roderick <sip:9910001@192.168.1.27>
Contact: "Roderick" <sip:9910001@192.168.1.104:5060>
Call-ID: 100FD5584507403DA28BC927854DD24F@192.168.1.27
CSeq: 27962 REGISTER
Expires: 60
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.1.104 : 5060 (NAT)
<--- Transmitting (no NAT) to 192.168.1.104:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK7FCA4317695A49B8ACB8F480819C5D02;received=192.168.1.104;rport=5060
From: Roderick <sip:9910001@192.168.1.27>;tag=1848079261
To: Roderick <sip:9910001@192.168.1.27>
Call-ID: 100FD5584507403DA28BC927854DD24F@192.168.1.27
CSeq: 27962 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:9910001@192.168.1.27:5065>
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 192.168.1.104:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK7FCA4317695A49B8ACB8F480819C5D02;received=192.168.1.104;rport=5060
From: Roderick <sip:9910001@192.168.1.27>;tag=1848079261
To: Roderick <sip:9910001@192.168.1.27>;tag=as37123fdd
Call-ID: 100FD5584507403DA28BC927854DD24F@192.168.1.27
CSeq: 27962 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 60
Contact: <sip:9910001@192.168.1.104:5060>;expires=60
Date: Tue, 26 Jun 2007 19:38:07 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '100FD5584507403DA28BC927854DD24F@192.168.1.27' in 32000 ms (Method: REGISTER)
Really destroying SIP dialog '306fcaeb1fe9e2e3@192.168.1.100' Method: REGISTER
asteriskRecorder*CLI>
<--- SIP read from 192.168.1.27:5060 --->
<------------->
--- (0 headers 0 lines) Nat keepalive ---
asteriskRecorder*CLI>
<--- SIP read from 192.168.1.100:5060 --->
REGISTER sip:192.168.1.27:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK4453d80c67348e02
From: <sip:279@192.168.1.27:5065>;tag=7ac086505cddff9f
To: <sip:279@192.168.1.27:5065>
Contact: <sip:279@192.168.1.100>
Supported: replaces
Call-ID: 306fcaeb1fe9e2e3@192.168.1.100
CSeq: 242 REGISTER
Expires: 60
User-Agent: Grandstream HT286 1.0.7.19
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
<------------->r*CLI>
--- (13 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.1.100 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.1.100:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK4453d80c67348e02;received=192.168.1.100
From: <sip:279@192.168.1.27:5065>;tag=7ac086505cddff9f
To: <sip:279@192.168.1.27:5065>
Call-ID: 306fcaeb1fe9e2e3@192.168.1.100
CSeq: 242 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:279@192.168.1.27:5065>
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 192.168.1.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.100;branch=z9hG4bK4453d80c67348e02;received=192.168.1.100
From: <sip:279@192.168.1.27:5065>;tag=7ac086505cddff9f
To: <sip:279@192.168.1.27:5065>;tag=as080494b8
Call-ID: 306fcaeb1fe9e2e3@192.168.1.100
CSeq: 242 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 60
Contact: <sip:279@192.168.1.100>;expires=60
Date: Tue, 26 Jun 2007 19:38:23 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '306fcaeb1fe9e2e3@192.168.1.100' in 32000 ms (Method: REGISTER)
Really destroying SIP dialog '1A58AF22A0D97CE7EDD9112B17A83AF3@192.168.1.27' Method: REGISTER
Really destroying SIP dialog '100FD5584507403DA28BC927854DD24F@192.168.1.27' Method: REGISTER
asteriskRecorder*CLI>
<--- SIP read from 192.168.1.104:5060 --->
<------------->
--- (0 headers 0 lines) Nat keepalive ---
asteriskRecorder*CLI>