rendered paste body*CLI> Reliably Transmitting (no NAT) to 10.0.2.201:5060:
OPTIONS sip:jason@10.0.2.201:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.2.231:5060;branch=z9hG4bK6cf925c5;rport
From: "asterisk" <sip:asterisk@10.0.2.231>;tag=as36918c0a
To: <sip:jason@10.0.2.201:5060>
Contact: <sip:asterisk@10.0.2.231>
Call-ID: 76b6b5287d554a9825f52ba570884595@10.0.2.231
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 01 Feb 2007 03:11:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
<--- SIP read from 10.0.2.201:5060 --->
SIP/2.0 200 OK
To: <sip:jason@10.0.2.201:5060>;tag=3ca707916c62eb32i0
From: "asterisk" <sip:asterisk@10.0.2.231>;tag=as36918c0a
Call-ID: 76b6b5287d554a9825f52ba570884595@10.0.2.231
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 10.0.2.231:5060;branch=z9hG4bK6cf925c5
Server: Linksys/SPA942-5.1.5
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '76b6b5287d554a9825f52ba570884595@10.0.2.231' Method: OPTIONS
Reliably Transmitting (NAT) to 203.2.134.1:5060:
OPTIONS sip:sip.internode.on.net SIP/2.0
Via: SIP/2.0/UDP 10.0.2.231:5060;branch=z9hG4bK483b77e1;rport
From: "asterisk" <sip:asterisk@10.0.2.231>;tag=as1121982d
To: <sip:sip.internode.on.net>
Contact: <sip:asterisk@10.0.2.231>
Call-ID: 24f5716a19ee682615492ca4625a01cb@10.0.2.231
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 01 Feb 2007 03:11:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
<--- SIP read from 203.2.134.1:5060 --->
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 10.0.2.231:5060;received=59.167.244.232;branch=z9hG4bK483b77e1;rport=1088
From: "asterisk" <sip:asterisk@10.0.2.231>;tag=as1121982d
To: <sip:sip.internode.on.net>;tag=aprqngfrt-2ed8f700000c6
Call-ID: 24f5716a19ee682615492ca4625a01cb@10.0.2.231
CSeq: 102 OPTIONS
Allow: INVITE,ACK,BYE,REGISTER,CANCEL,PRACK,INFO,SUBSCRIBE,NOTIFY,UPDATE
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '24f5716a19ee682615492ca4625a01cb@10.0.2.231' Method: OPTIONS
<--- SIP read from 10.0.2.201:5060 --->
INVITE sip:2@10.0.2.231 SIP/2.0
Via: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-1d940943
From: "Jason" <sip:jason@10.0.2.231>;tag=61e2bb0f5a8e7e1bo0
To: <sip:2@10.0.2.231>
Call-ID: 3e96251f-b51b9b4b@10.0.2.201
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "Jason" <sip:jason@10.0.2.201:5060>
Expires: 240
User-Agent: Linksys/SPA942-5.1.5
Content-Length: 393
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 7791456 7791456 IN IP4 10.0.2.201
s=-
c=IN IP4 10.0.2.201
t=0 0
m=audio 16394 RTP/AVP 18 0 2 4 8 96 97 98 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (14 headers 18 lines) ---
Sending to 10.0.2.201 : 5060 (no NAT)
Using INVITE request as basis request - 3e96251f-b51b9b4b@10.0.2.201
<--- Reliably Transmitting (no NAT) to 10.0.2.201:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-1d940943;received=10.0.2.201
From: "Jason" <sip:jason@10.0.2.231>;tag=61e2bb0f5a8e7e1bo0
To: <sip:2@10.0.2.231>;tag=as68029e8d
Call-ID: 3e96251f-b51b9b4b@10.0.2.201
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="sip.internode.on.net", nonce="66f141e1"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '3e96251f-b51b9b4b@10.0.2.201' in 32000 ms (Method: INVITE)
Found user 'jason'
<--- SIP read from 10.0.2.201:5060 --->
ACK sip:2@10.0.2.231 SIP/2.0
Via: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-1d940943
From: "Jason" <sip:jason@10.0.2.231>;tag=61e2bb0f5a8e7e1bo0
To: <sip:2@10.0.2.231>;tag=as68029e8d
Call-ID: 3e96251f-b51b9b4b@10.0.2.201
CSeq: 101 ACK
Max-Forwards: 70
Contact: "Jason" <sip:jason@10.0.2.201:5060>
User-Agent: Linksys/SPA942-5.1.5
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from 10.0.2.201:5060 --->
INVITE sip:2@10.0.2.231 SIP/2.0
Via: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-fcb1c6bb
From: "Jason" <sip:jason@10.0.2.231>;tag=61e2bb0f5a8e7e1bo0
To: <sip:2@10.0.2.231>
Call-ID: 3e96251f-b51b9b4b@10.0.2.201
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username="jason",realm="sip.internode.on.net",nonce="66f141e1",uri="sip:2@10.0.2.231",algorithm=MD5,response="7401b0554c9c8132263bfeca8e3a96de"
Contact: "Jason" <sip:jason@10.0.2.201:5060>
Expires: 240
User-Agent: Linksys/SPA942-5.1.5
Content-Length: 393
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 7791456 7791456 IN IP4 10.0.2.201
s=-
c=IN IP4 10.0.2.201
t=0 0
m=audio 16394 RTP/AVP 18 0 2 4 8 96 97 98 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (15 headers 18 lines) ---
Sending to 10.0.2.201 : 5060 (no NAT)
Using INVITE request as basis request - 3e96251f-b51b9b4b@10.0.2.201
Found user 'jason'
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 101
Peer audio RTP is at port 10.0.2.201:16394
Found description format G729a for ID 18
Found description format PCMU for ID 0
Found description format G726-32 for ID 2
Found description format G723 for ID 4
Found description format PCMA for ID 8
Found description format G726-40 for ID 96
Found description format G726-24 for ID 97
Found description format G726-16 for ID 98
Found description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.0.2.201:16394
Looking for 2 in myphones (domain 10.0.2.231)
list_route: hop: <sip:jason@10.0.2.201:5060>
<--- Transmitting (no NAT) to 10.0.2.201:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-fcb1c6bb;received=10.0.2.201
From: "Jason" <sip:jason@10.0.2.231>;tag=61e2bb0f5a8e7e1bo0
To: <sip:2@10.0.2.231>
Call-ID: 3e96251f-b51b9b4b@10.0.2.201
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2@10.0.2.231>
Content-Length: 0
<------------>
-- Executing [2@myphones:1] Dial("SIP/jason-081d7cf8", "SIP/nodephone/0871271204") in new stack
Audio is at 10.0.2.231 port 15596
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 203.2.134.1:5060:
INVITE sip:0871271204@sip.internode.on.net SIP/2.0
Via: SIP/2.0/UDP 10.0.2.231:5060;branch=z9hG4bK2e1a1505;rport
From: "Jason" <sip:0290432296@sip.internode.on.net>;tag=as7774e54b
To: <sip:0871271204@sip.internode.on.net>
Contact: <sip:0290432296@10.0.2.231>
Call-ID: 4b0247f9697a5651296659876c7edbae@sip.internode.on.net
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Jason" <sip:jason@sip.internode.on.net>;privacy=off;screen=no
Date: Thu, 01 Feb 2007 03:11:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 234
v=0
o=root 6406 6406 IN IP4 10.0.2.231
s=session
c=IN IP4 10.0.2.231
t=0 0
m=audio 15596 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
---
-- Called nodephone/0871271204
<--- SIP read from 203.2.134.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.2.231:5060;received=59.167.244.232;branch=z9hG4bK2e1a1505;rport=1088
From: "Jason" <sip:0290432296@sip.internode.on.net>;tag=as7774e54b
To: <sip:0871271204@sip.internode.on.net>
Call-ID: 4b0247f9697a5651296659876c7edbae@sip.internode.on.net
CSeq: 102 INVITE
<------------->
--- (6 headers 0 lines) ---
<--- SIP read from 203.2.134.1:5060 --->
SIP/2.0 408 Request Timeout
Via: SIP/2.0/UDP 10.0.2.231:5060;received=59.167.244.232;branch=z9hG4bK2e1a1505;rport=1088
From: "Jason" <sip:0290432296@sip.internode.on.net>;tag=as7774e54b
To: <sip:0871271204@sip.internode.on.net>;tag=aprqngfrt-10mq6t30000c6
Call-ID: 4b0247f9697a5651296659876c7edbae@sip.internode.on.net
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
-- Got SIP response 408 "Request Timeout" back from 203.2.134.1
Transmitting (NAT) to 203.2.134.1:5060:
ACK sip:0871271204@sip.internode.on.net SIP/2.0
Via: SIP/2.0/UDP 10.0.2.231:5060;branch=z9hG4bK2e1a1505;rport
From: "Jason" <sip:0290432296@sip.internode.on.net>;tag=as7774e54b
To: <sip:0871271204@sip.internode.on.net>;tag=aprqngfrt-10mq6t30000c6
Contact: <sip:0290432296@10.0.2.231>
Call-ID: 4b0247f9697a5651296659876c7edbae@sip.internode.on.net
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Jason" <sip:jason@sip.internode.on.net>;privacy=off;screen=no
Content-Length: 0
---
== Everyone is busy/congested at this time (1:0/0/1)
Really destroying SIP dialog '4b0247f9697a5651296659876c7edbae@sip.internode.on.net' Method: INVITE
<--- SIP read from 10.0.2.201:5060 --->
CANCEL sip:2@10.0.2.231 SIP/2.0
Via: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-fcb1c6bb
From: "Jason" <sip:jason@10.0.2.231>;tag=61e2bb0f5a8e7e1bo0
To: <sip:2@10.0.2.231>
Call-ID: 3e96251f-b51b9b4b@10.0.2.201
CSeq: 102 CANCEL
Max-Forwards: 70
Proxy-Authorization: Digest username="jason",realm="sip.internode.on.net",nonce="66f141e1",uri="sip:2@10.0.2.231",algorithm=MD5,response="ab6cae345fa4ae0667a4542c59d840e3"
User-Agent: Linksys/SPA942-5.1.5
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 10.0.2.201 : 5060 (no NAT)
<--- Reliably Transmitting (no NAT) to 10.0.2.201:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-fcb1c6bb;received=10.0.2.201
From: "Jason" <sip:jason@10.0.2.231>;tag=61e2bb0f5a8e7e1bo0
To: <sip:2@10.0.2.231>;tag=as139af391
Call-ID: 3e96251f-b51b9b4b@10.0.2.201
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 10.0.2.201:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-fcb1c6bb;received=10.0.2.201
From: "Jason" <sip:jason@10.0.2.231>;tag=61e2bb0f5a8e7e1bo0
To: <sip:2@10.0.2.231>;tag=as139af391
Call-ID: 3e96251f-b51b9b4b@10.0.2.201
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:2@10.0.2.231>
Content-Length: 0
<------------>
<--- SIP read from 10.0.2.201:5060 --->
ACK sip:2@10.0.2.231 SIP/2.0
Via: SIP/2.0/UDP 10.0.2.201:5060;branch=z9hG4bK-fcb1c6bb
From: "Jason" <sip:jason@10.0.2.231>;tag=61e2bb0f5a8e7e1bo0
To: <sip:2@10.0.2.231>;tag=as139af391
Call-ID: 3e96251f-b51b9b4b@10.0.2.201
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="jason",realm="sip.internode.on.net",nonce="66f141e1",uri="sip:2@10.0.2.231",algorithm=MD5,response="246adcfd5d54d712e675a63fcad7b061"
Contact: "Jason" <sip:jason@10.0.2.201:5060>
User-Agent: Linksys/SPA942-5.1.5
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '3e96251f-b51b9b4b@10.0.2.201' Method: ACK