rendered paste body<--- SIP read from 192.168.11.125:5060 --->
INVITE sip:214057@192.168.11.124:5060;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via:SIP/2.0/UDP 192.168.11.125:5060;branch=z9hG4bK1e405bf876bfdb1b
From:"214058" <sip:214058@192.168.11.124:5060>;tag=996ccafd-685050
To:<sip:214057@192.168.11.124:5060;user=phone>
Call-ID:D1B9-1C8E-466850504B766C9D6E47-003@SipHost
CSeq:7 INVITE
Contact:<sip:214058@192.168.11.125:5060>
Expires:90
Max-Forwards:70
Remote-Party-ID:<sip:214058@192.168.11.124:5060>;party=calling;privacy=off;screen=yes
Supported:replaces
User-Agent:dlink 12-36-7924821
Content-Type:application/sdp
Content-Length:154
v=0
o=214058 1792244960 1792244960 IN IP4 192.168.11.125
s=Session SDP
c=IN IP4 192.168.11.125
t=0 0
m=audio 9000 RTP/AVP 8
a=rtpmap:8 PCMA/8000/1
<------------->
--- (15 headers 7 lines) ---
Sending to 192.168.11.125 : 5060 (no NAT)
Using INVITE request as basis request - D1B9-1C8E-466850504B766C9D6E47-003@SipHost
bee*CLI>
<--- Reliably Transmitting (no NAT) to 192.168.11.125:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.11.125:5060;branch=z9hG4bK1e405bf876bfdb1b;received=192.168.11.125
From: "214058" <sip:214058@192.168.11.124:5060>;tag=996ccafd-685050
To: <sip:214057@192.168.11.124:5060;user=phone>;tag=as05742a67
Call-ID: D1B9-1C8E-466850504B766C9D6E47-003@SipHost
CSeq: 7 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="195bec50"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'D1B9-1C8E-466850504B766C9D6E47-003@SipHost' in 32000 ms (Method: INVITE)
Found user '214058'
bee*CLI>
<--- SIP read from 192.168.11.125:5060 --->
ACK sip:214057@192.168.11.124:5060;user=phone SIP/2.0
Via:SIP/2.0/UDP 192.168.11.125:5060;branch=z9hG4bK1e405bf876bfdb1b
From:"214058" <sip:214058@192.168.11.124:5060>;tag=996ccafd-685050
To:<sip:214057@192.168.11.124:5060;user=phone>;tag=as05742a67
Call-ID:D1B9-1C8E-466850504B766C9D6E47-003@SipHost
CSeq:7 ACK
Max-Forwards:70
Content-Length:0
<------------->
--- (8 headers 0 lines) ---
bee*CLI>
<--- SIP read from 192.168.11.125:5060 --->
INVITE sip:214057@192.168.11.124:5060;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via:SIP/2.0/UDP 192.168.11.125:5060;branch=z9hG4bK6b90dd9eecb34952
From:"214058" <sip:214058@192.168.11.124:5060>;tag=996ccafd-685050
To:<sip:214057@192.168.11.124:5060;user=phone>
Call-ID:D1B9-1C8E-466850504B766C9D6E47-003@SipHost
CSeq:8 INVITE
Contact:<sip:214058@192.168.11.125:5060>
Expires:90
Max-Forwards:70
Proxy-Authorization:Digest username="214058",realm="asterisk",nonce="195bec50",uri="sip:214057@192.168.11.124:5060;user=phone",response="b4569e17f4320ddb246e8ca2de512ab0",algorithm=MD5
Remote-Party-ID:<sip:214058@192.168.11.124:5060>;party=calling;privacy=off;screen=yes
Supported:replaces
User-Agent:dlink 12-36-7924821
Content-Type:application/sdp
Content-Length:154
v=0
o=214058 1792244960 1792244960 IN IP4 192.168.11.125
s=Session SDP
c=IN IP4 192.168.11.125
t=0 0
m=audio 9000 RTP/AVP 8
a=rtpmap:8 PCMA/8000/1
<------------->
--- (16 headers 7 lines) ---
Sending to 192.168.11.125 : 5060 (no NAT)
Using INVITE request as basis request - D1B9-1C8E-466850504B766C9D6E47-003@SipHost
Found user '214058'
Found RTP audio format 8
Peer audio RTP is at port 192.168.11.125:9000
Found description format PCMA for ID 8
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.11.125:9000
Looking for 214057 in local-phones (domain 192.168.11.124)
list_route: hop: <sip:214058@192.168.11.125:5060>
<--- Transmitting (no NAT) to 192.168.11.125:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.11.125:5060;branch=z9hG4bK6b90dd9eecb34952;received=192.168.11.125
From: "214058" <sip:214058@192.168.11.124:5060>;tag=996ccafd-685050
To: <sip:214057@192.168.11.124:5060;user=phone>
Call-ID: D1B9-1C8E-466850504B766C9D6E47-003@SipHost
CSeq: 8 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:214057@192.168.11.124>
Content-Length: 0
<------------>
bee*CLI>
<--- Transmitting (no NAT) to 192.168.11.125:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.11.125:5060;branch=z9hG4bK6b90dd9eecb34952;received=192.168.11.125
From: "214058" <sip:214058@192.168.11.124:5060>;tag=996ccafd-685050
To: <sip:214057@192.168.11.124:5060;user=phone>;tag=as09799ece
Call-ID: D1B9-1C8E-466850504B766C9D6E47-003@SipHost
CSeq: 8 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:214057@192.168.11.124>
Content-Length: 0
<------------>
Audio is at 192.168.11.124 port 14902
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.11.125:5060:
INVITE sip:214058@192.168.11.125:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK4288ba1a;rport
From: "Unknown" <sip:Unknown@192.168.11.124>;tag=as73d2b605
To: <sip:214058@192.168.11.125:5060>
Contact: <sip:Unknown@192.168.11.124>
Call-ID: 38044d8212576c61228fe12265417374@192.168.11.124
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 16 Jul 2007 08:25:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 244
v=0
o=root 20154 20154 IN IP4 192.168.11.124
s=session
c=IN IP4 192.168.11.124
t=0 0
m=audio 14902 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
bee*CLI>
<--- SIP read from 192.168.11.125:5060 --->
SIP/2.0 100 Trying
Via:SIP/2.0/UDP 192.168.11.124:5060;rport;branch=z9hG4bK4288ba1a
From:"Unknown" <sip:Unknown@192.168.11.124>;tag=as73d2b605
To:<sip:214058@192.168.11.125:5060>
Call-ID:38044d8212576c61228fe12265417374@192.168.11.124
CSeq:102 INVITE
Content-Type:application/sdp
Content-Length:0
<------------->
--- (8 headers 0 lines) ---
bee*CLI>
<--- SIP read from 192.168.11.125:5060 --->
SIP/2.0 486 Busy here
Via:SIP/2.0/UDP 192.168.11.124:5060;rport;branch=z9hG4bK4288ba1a
From:"Unknown" <sip:Unknown@192.168.11.124>;tag=as73d2b605
To:<sip:214058@192.168.11.125:5060>;tag=332065da-685052
Call-ID:38044d8212576c61228fe12265417374@192.168.11.124
CSeq:102 INVITE
Contact:<sip:214058@192.168.11.125:5060>
User-Agent:dlink 12-36-7924821
Content-Length:0
<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 192.168.11.125:5060:
ACK sip:214058@192.168.11.125:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.11.124:5060;branch=z9hG4bK4288ba1a;rport
From: "Unknown" <sip:Unknown@192.168.11.124>;tag=as73d2b605
To: <sip:214058@192.168.11.125:5060>;tag=332065da-685052
Contact: <sip:Unknown@192.168.11.124>
Call-ID: 38044d8212576c61228fe12265417374@192.168.11.124
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Really destroying SIP dialog '38044d8212576c61228fe12265417374@192.168.11.124' Method: INVITE
Audio is at 192.168.11.124 port 17714
Adding codec 0x8 (alaw) to SDP
bee*CLI>
<--- Reliably Transmitting (no NAT) to 192.168.11.125:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.11.125:5060;branch=z9hG4bK6b90dd9eecb34952;received=192.168.11.125
From: "214058" <sip:214058@192.168.11.124:5060>;tag=996ccafd-685050
To: <sip:214057@192.168.11.124:5060;user=phone>;tag=as09799ece
Call-ID: D1B9-1C8E-466850504B766C9D6E47-003@SipHost
CSeq: 8 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:214057@192.168.11.124>
Content-Type: application/sdp
Content-Length: 188
v=0
o=root 20154 20154 IN IP4 192.168.11.124
s=session
c=IN IP4 192.168.11.124
t=0 0
m=audio 17714 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
bee*CLI>
<--- SIP read from 192.168.11.125:5060 --->
ACK sip:214057@192.168.11.124 SIP/2.0
Via:SIP/2.0/UDP 192.168.11.125:5060;branch=z9hG4bKefbfd298bebe73f2
From:"214058" <sip:214058@192.168.11.124:5060>;tag=996ccafd-685050
To:<sip:214057@192.168.11.124:5060;user=phone>;tag=as09799ece
Call-ID:D1B9-1C8E-466850504B766C9D6E47-003@SipHost
CSeq:8 ACK
Max-Forwards:70
Proxy-Authorization:Digest username="214058",realm="asterisk",nonce="195bec50",uri="sip:214057@192.168.11.124:5060;user=phone",response="b4569e17f4320ddb246e8ca2de512ab0",algorithm=MD5
User-Agent:dlink 12-36-7924821
Content-Length:0
<------------->
--- (10 headers 0 lines) ---
[Jul 16 12:26:07] WARNING[20205]: rtp.c:885 ast_rtcp_read: RTCP Read too short
bee*CLI>
<--- SIP read from 192.168.11.125:5060 --->
BYE sip:214057@192.168.11.124 SIP/2.0
Via:SIP/2.0/UDP 192.168.11.125:5060;branch=z9hG4bK0cee6af45125ff96
From:"214058" <sip:214058@192.168.11.124:5060>;tag=996ccafd-685050
To:<sip:214057@192.168.11.124:5060;user=phone>;tag=as09799ece
Call-ID:D1B9-1C8E-466850504B766C9D6E47-003@SipHost
CSeq:9 BYE
Max-Forwards:70
User-Agent:dlink 12-36-7924821
Content-Length:0
<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.11.125 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.11.125:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.11.125:5060;branch=z9hG4bK0cee6af45125ff96;received=192.168.11.125
From: "214058" <sip:214058@192.168.11.124:5060>;tag=996ccafd-685050
To: <sip:214057@192.168.11.124:5060;user=phone>;tag=as09799ece
Call-ID: D1B9-1C8E-466850504B766C9D6E47-003@SipHost
CSeq: 9 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:214057@192.168.11.124>
Content-Length: 0
<------------>
Really destroying SIP dialog 'D1B9-1C8E-466850504B766C9D6E47-003@SipHost' Method: BYE