rendered paste body14:01:29.862830 IP (tos 0xbf,CE, ttl 64, id 48379, offset 0, flags [none], proto: UDP (17), length: 992) 10.30.0.1.5060 > 192.168.101.2.5060: SIP, length: 964
INVITE sip:107@10.30.0.1:5060;user=phone SIP/2.0
Record-Route: <sip:10.30.0.1;lr=on;ftag=665B56FF-29E48C66>
Via: SIP/2.0/UDP 10.30.0.1;branch=z9hG4bK4f6a.7e4f0e41.0
Via: SIP/2.0/UDP 10.30.0.201;branch=z9hG4bK45e74a31E941F2B2
From: "Richard Metivier" <sip:201@10.30.0.1>;tag=665B56FF-29E48C66
To: <sip:107@10.30.0.1;user=phone>
CSeq: 1 INVITE
Call-ID: 36793eb-a825f85-2fa1121c@10.30.0.201
Contact: <sip:201@10.30.0.201>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 246
v=0
o=- 978316398 978316398 IN IP4 10.30.0.201
s=Polycom IP Phone
c=IN IP4 10.30.0.201
t=0 0
a=sendrecv
m=audio 10004 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
14:01:29.865375 IP (tos 0xb8, ttl 63, id 58992, offset 0, flags [none], proto: UDP (17), length: 524) 192.168.101.2.5060 > 10.30.0.1.5060: SIP, length: 496
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.30.0.1;branch=z9hG4bK4f6a.7e4f0e41.0;received=10.30.0.1
Via: SIP/2.0/UDP 10.30.0.201;branch=z9hG4bK45e74a31E941F2B2
From: "Richard Metivier" <sip:201@10.30.0.1>;tag=665B56FF-29E48C66
To: <sip:107@10.30.0.1;user=phone>
Call-ID: 36793eb-a825f85-2fa1121c@10.30.0.201
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:107@207.134.166.34>
Content-Length: 0
14:01:30.134817 IP (tos 0xb8, ttl 63, id 58993, offset 0, flags [none], proto: UDP (17), length: 540) 192.168.101.2.5060 > 10.30.0.1.5060: SIP, length: 512
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.30.0.1;branch=z9hG4bK4f6a.7e4f0e41.0;received=10.30.0.1
Via: SIP/2.0/UDP 10.30.0.201;branch=z9hG4bK45e74a31E941F2B2
From: "Richard Metivier" <sip:201@10.30.0.1>;tag=665B56FF-29E48C66
To: <sip:107@10.30.0.1;user=phone>;tag=as7555abb2
Call-ID: 36793eb-a825f85-2fa1121c@10.30.0.201
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:107@207.134.166.34>
Content-Length: 0
14:01:34.416289 IP (tos 0xb8, ttl 63, id 58994, offset 0, flags [none], proto: UDP (17), length: 896) 192.168.101.2.5060 > 10.30.0.1.5060: SIP, length: 868
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.30.0.1;branch=z9hG4bK4f6a.7e4f0e41.0;received=10.30.0.1
Via: SIP/2.0/UDP 10.30.0.201;branch=z9hG4bK45e74a31E941F2B2
Record-Route: <sip:10.30.0.1;lr=on;ftag=665B56FF-29E48C66>
From: "Richard Metivier" <sip:201@10.30.0.1>;tag=665B56FF-29E48C66
To: <sip:107@10.30.0.1;user=phone>;tag=as7555abb2
Call-ID: 36793eb-a825f85-2fa1121c@10.30.0.201
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:107@207.134.166.34>
Content-Type: application/sdp
Content-Length: 268
v=0
o=root 13185 13185 IN IP4 207.134.166.34
s=session
c=IN IP4 207.134.166.34
t=0 0
m=audio 16376 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
14:01:34.460791 IP (tos 0xbf,CE, ttl 64, id 48387, offset 0, flags [none], proto: UDP (17), length: 668) 10.30.0.1.5060 > 192.168.101.2.5060: SIP, length: 640
ACK sip:107@207.134.166.34 SIP/2.0
Record-Route: <sip:10.30.0.1;lr=on;ftag=665B56FF-29E48C66>
Via: SIP/2.0/UDP 10.30.0.1;branch=z9hG4bK4f6a.7e4f0e41.2
Via: SIP/2.0/UDP 10.30.0.201;branch=z9hG4bK5c9d04595187D99A
From: "Richard Metivier" <sip:201@10.30.0.1>;tag=665B56FF-29E48C66
To: <sip:107@10.30.0.1;user=phone>;tag=as7555abb2
CSeq: 1 ACK
Call-ID: 36793eb-a825f85-2fa1121c@10.30.0.201
Contact: <sip:201@10.30.0.201>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098
Max-Forwards: 69
Content-Length: 0
P-hint: rr-enforced
14:01:34.463489 IP (tos 0xb8, ttl 63, id 58995, offset 0, flags [none], proto: UDP (17), length: 914) 192.168.101.2.5060 > 10.30.0.1.5060: SIP, length: 886
INVITE sip:201@10.30.0.201 SIP/2.0
Via: SIP/2.0/UDP 207.134.166.34:5060;branch=z9hG4bK52e31b40;rport
Route: <sip:10.30.0.1;lr=on;ftag=665B56FF-29E48C66>
From: <sip:107@10.30.0.1;user=phone>;tag=as7555abb2
To: "Richard Metivier" <sip:201@10.30.0.1>;tag=665B56FF-29E48C66
Contact: <sip:107@207.134.166.34>
Call-ID: 36793eb-a825f85-2fa1121c@10.30.0.201
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: 270
v=0
o=root 13185 13186 IN IP4 192.168.101.198
s=session
c=IN IP4 192.168.101.198
t=0 0
m=audio 10012 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
14:01:34.463754 IP (tos 0xbf,CE, ttl 64, id 48389, offset 0, flags [none], proto: UDP (17), length: 400) 10.30.0.1.5060 > 192.168.101.2.5060: SIP, length: 372
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 207.134.166.34:5060;branch=z9hG4bK52e31b40;rport=5060;received=192.168.101.2
From: <sip:107@10.30.0.1;user=phone>;tag=as7555abb2
To: "Richard Metivier" <sip:201@10.30.0.1>;tag=665B56FF-29E48C66
Call-ID: 36793eb-a825f85-2fa1121c@10.30.0.201
CSeq: 102 INVITE
Server: OpenSER (1.2.0-notls (i386/freebsd))
Content-Length: 0
14:01:34.463811 IP (tos 0xbf,CE, ttl 64, id 48390, offset 0, flags [none], proto: UDP (17), length: 1021) 10.30.0.1.5060 > 192.168.101.2.5060: SIP, length: 993
INVITE sip:201@10.30.0.201 SIP/2.0
Record-Route: <sip:10.30.0.1;lr=on;ftag=as7555abb2>
Via: SIP/2.0/UDP 10.30.0.1;branch=z9hG4bKc85c.bd01d981.0
Via: SIP/2.0/UDP 207.134.166.34:5060;received=192.168.101.2;branch=z9hG4bK52e31b40;rport=5060
From: <sip:107@10.30.0.1;user=phone>;tag=as7555abb2
To: "Richard Metivier" <sip:201@10.30.0.1>;tag=665B56FF-29E48C66
Contact: <sip:107@207.134.166.34>
Call-ID: 36793eb-a825f85-2fa1121c@10.30.0.201
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 69
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: 270
P-hint: rr-enforced
v=0
o=root 13185 13186 IN IP4 192.168.101.198
s=session
c=IN IP4 192.168.101.198
t=0 0
m=audio 10012 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
14:01:34.466242 IP (tos 0xb8, ttl 63, id 58996, offset 0, flags [none], proto: UDP (17), length: 622) 192.168.101.2.5060 > 10.30.0.1.5060: SIP, length: 594
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 10.30.0.1;branch=z9hG4bKc85c.bd01d981.0;received=10.30.0.1
Via: SIP/2.0/UDP 207.134.166.34:5060;received=192.168.101.2;branch=z9hG4bK52e31b40;rport=5060
From: <sip:107@10.30.0.1;user=phone>;tag=as7555abb2
To: "Richard Metivier" <sip:201@10.30.0.1>;tag=665B56FF-29E48C66
Call-ID: 36793eb-a825f85-2fa1121c@10.30.0.201
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
14:01:34.466359 IP (tos 0xbf,CE, ttl 64, id 48392, offset 0, flags [none], proto: UDP (17), length: 424) 10.30.0.1.5060 > 192.168.101.2.5060: SIP, length: 396
ACK sip:201@10.30.0.201 SIP/2.0
Via: SIP/2.0/UDP 10.30.0.1;branch=z9hG4bKc85c.bd01d981.0
From: <sip:107@10.30.0.1;user=phone>;tag=as7555abb2
Call-ID: 36793eb-a825f85-2fa1121c@10.30.0.201
To: "Richard Metivier" <sip:201@10.30.0.1>;tag=665B56FF-29E48C66
CSeq: 102 ACK
Route: <sip:10.30.0.1;lr=on;ftag=665B56FF-29E48C66>
User-Agent: OpenSER (1.2.0-notls (i386/freebsd))
Content-Length: 0
14:01:34.466397 IP (tos 0xbf,CE, ttl 64, id 48393, offset 0, flags [none], proto: UDP (17), length: 545) 10.30.0.1.5060 > 192.168.101.2.5060: SIP, length: 517
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 207.134.166.34:5060;received=192.168.101.2;branch=z9hG4bK52e31b40;rport=5060
From: <sip:107@10.30.0.1;user=phone>;tag=as7555abb2
To: "Richard Metivier" <sip:201@10.30.0.1>;tag=665B56FF-29E48C66
Call-ID: 36793eb-a825f85-2fa1121c@10.30.0.201
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
14:01:34.467518 IP (tos 0xb8, ttl 63, id 58997, offset 0, flags [none], proto: UDP (17), length: 462) 192.168.101.2.5060 > 10.30.0.1.5060: SIP, length: 434
ACK sip:201@10.30.0.201 SIP/2.0
Via: SIP/2.0/UDP 207.134.166.34:5060;branch=z9hG4bK52e31b40;rport
Route: <sip:10.30.0.1;lr=on;ftag=665B56FF-29E48C66>
From: <sip:107@10.30.0.1;user=phone>;tag=as7555abb2
To: "Richard Metivier" <sip:201@10.30.0.1>;tag=665B56FF-29E48C66
Contact: <sip:107@207.134.166.34>
Call-ID: 36793eb-a825f85-2fa1121c@10.30.0.201
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
14:01:34.491051 IP (tos 0xb8, ttl 63, id 58998, offset 0, flags [none], proto: UDP (17), length: 427) 192.168.101.2.5060 > 10.30.0.1.5060: SIP, length: 399
BYE sip:201@10.30.0.201 SIP/2.0
Via: SIP/2.0/UDP 207.134.166.34:5060;branch=z9hG4bK4b779266;rport
Route: <sip:10.30.0.1;lr=on;ftag=665B56FF-29E48C66>
From: <sip:107@10.30.0.1;user=phone>;tag=as7555abb2
To: "Richard Metivier" <sip:201@10.30.0.1>;tag=665B56FF-29E48C66
Call-ID: 36793eb-a825f85-2fa1121c@10.30.0.201
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
14:01:34.491312 IP (tos 0xbf,CE, ttl 64, id 48395, offset 0, flags [none], proto: UDP (17), length: 534) 10.30.0.1.5060 > 192.168.101.2.5060: SIP, length: 506
BYE sip:201@10.30.0.201 SIP/2.0
Record-Route: <sip:10.30.0.1;lr=on;ftag=as7555abb2>
Via: SIP/2.0/UDP 10.30.0.1;branch=z9hG4bKd85c.c76adbf6.0
Via: SIP/2.0/UDP 207.134.166.34:5060;received=192.168.101.2;branch=z9hG4bK4b779266;rport=5060
From: <sip:107@10.30.0.1;user=phone>;tag=as7555abb2
To: "Richard Metivier" <sip:201@10.30.0.1>;tag=665B56FF-29E48C66
Call-ID: 36793eb-a825f85-2fa1121c@10.30.0.201
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0
P-hint: rr-enforced
14:01:34.493272 IP (tos 0xb8, ttl 63, id 58999, offset 0, flags [none], proto: UDP (17), length: 621) 192.168.101.2.5060 > 10.30.0.1.5060: SIP, length: 593
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.30.0.1;branch=z9hG4bKd85c.c76adbf6.0;received=10.30.0.1
Via: SIP/2.0/UDP 207.134.166.34:5060;received=192.168.101.2;branch=z9hG4bK4b779266;rport=5060
Record-Route: <sip:10.30.0.1;lr=on;ftag=as7555abb2>
From: <sip:107@10.30.0.1;user=phone>;tag=as7555abb2
To: "Richard Metivier" <sip:201@10.30.0.1>;tag=665B56FF-29E48C66
Call-ID: 36793eb-a825f85-2fa1121c@10.30.0.201
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:107@207.134.166.34>
Content-Length: 0
14:01:34.493391 IP (tos 0xbf,CE, ttl 64, id 48397, offset 0, flags [none], proto: UDP (17), length: 544) 10.30.0.1.5060 > 192.168.101.2.5060: SIP, length: 516
SIP/2.0 200 OK
Via: SIP/2.0/UDP 207.134.166.34:5060;received=192.168.101.2;branch=z9hG4bK4b779266;rport=5060
Record-Route: <sip:10.30.0.1;lr=on;ftag=as7555abb2>
From: <sip:107@10.30.0.1;user=phone>;tag=as7555abb2
To: "Richard Metivier" <sip:201@10.30.0.1>;tag=665B56FF-29E48C66
Call-ID: 36793eb-a825f85-2fa1121c@10.30.0.201
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:107@207.134.166.34>
Content-Length: 0