rendered paste body<--- SIP read from 10.1.7.167:5060 --->
INVITE sip:1000@10.1.7.17 SIP/2.0
Via: SIP/2.0/UDP 10.1.7.167:5060;branch=z9hG4bK-e19c109d
From: "Hello Worldz" <sip:888@10.1.7.17>;tag=44c160be400f81ado0
To: <sip:1000@10.1.7.17>
Call-ID: cf21c6d2-e12232a9@10.1.7.167
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "Hello Worldz" <sip:888@10.1.7.167:5060>
Expires: 240
User-Agent: Sipura/SPA921-4.1.10(b)
Content-Length: 389
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp
v=0
o=- 92372 92372 IN IP4 10.1.7.167
s=-
c=IN IP4 10.1.7.167
t=0 0
m=audio 16480 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/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:20
a=sendrecv
<------------->
--- (13 headers 18 lines) ---
== Using TOS bits 0
== Using CoS mark 0
Sending to 10.1.7.167 : 5060 (no NAT)
Using INVITE request as basis request - cf21c6d2-e12232a9@10.1.7.167
No user '888' in SIP users list
Found peer '888' for '888' from 10.1.7.167:5060
<--- Reliably Transmitting (no NAT) to 10.1.7.167:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.7.167:5060;branch=z9hG4bK-e19c109d;received=10.1.7.167
From: "Hello Worldz" <sip:888@10.1.7.17>;tag=44c160be400f81ado0
To: <sip:1000@10.1.7.17>;tag=as1174f139
Call-ID: cf21c6d2-e12232a9@10.1.7.167
CSeq: 101 INVITE
User-Agent: Asterisk PBX SVN-trunk-r63330
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3f9e445f"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'cf21c6d2-e12232a9@10.1.7.167' in 32000 ms (Method: INVITE)
<--- SIP read from 10.1.7.167:5060 --->
ACK sip:1000@10.1.7.17 SIP/2.0
Via: SIP/2.0/UDP 10.1.7.167:5060;branch=z9hG4bK-e19c109d
From: "Hello Worldz" <sip:888@10.1.7.17>;tag=44c160be400f81ado0
To: <sip:1000@10.1.7.17>;tag=as1174f139
Call-ID: cf21c6d2-e12232a9@10.1.7.167
CSeq: 101 ACK
Max-Forwards: 70
Contact: "Hello Worldz" <sip:888@10.1.7.167:5060>
User-Agent: Sipura/SPA921-4.1.10(b)
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from 10.1.7.167:5060 --->
INVITE sip:1000@10.1.7.17 SIP/2.0
Via: SIP/2.0/UDP 10.1.7.167:5060;branch=z9hG4bK-42870f9d
From: "Hello Worldz" <sip:888@10.1.7.17>;tag=44c160be400f81ado0
To: <sip:1000@10.1.7.17>
Call-ID: cf21c6d2-e12232a9@10.1.7.167
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="888",realm="asterisk",nonce="3f9e445f",uri="sip:1000@10.1.7.17",algorithm=MD5,response="d2b7cfcff25a2e7fa7c3091167f3f19b"
Contact: "Hello Worldz" <sip:888@10.1.7.167:5060>
Expires: 240
User-Agent: Sipura/SPA921-4.1.10(b)
Content-Length: 389
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp
v=0
o=- 92372 92372 IN IP4 10.1.7.167
s=-
c=IN IP4 10.1.7.167
t=0 0
m=audio 16480 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/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:20
a=sendrecv
<------------->
--- (14 headers 18 lines) ---
Sending to 10.1.7.167 : 5060 (no NAT)
Using INVITE request as basis request - cf21c6d2-e12232a9@10.1.7.167
No user '888' in SIP users list
Found peer '888' for '888' from 10.1.7.167:5060
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
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 10.1.7.167:16480
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 G729a for ID 18
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 - 0x4 (ulaw), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.1.7.167:16480
Looking for 1000 in users (domain 10.1.7.17)
<--- Reliably Transmitting (no NAT) to 10.1.7.167:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.1.7.167:5060;branch=z9hG4bK-42870f9d;received=10.1.7.167
From: "Hello Worldz" <sip:888@10.1.7.17>;tag=44c160be400f81ado0
To: <sip:1000@10.1.7.17>;tag=as1174f139
Call-ID: cf21c6d2-e12232a9@10.1.7.167
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-trunk-r63330
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'cf21c6d2-e12232a9@10.1.7.167' in 32000 ms (Method: INVITE)
<--- SIP read from 10.1.7.167:5060 --->
ACK sip:1000@10.1.7.17 SIP/2.0
Via: SIP/2.0/UDP 10.1.7.167:5060;branch=z9hG4bK-42870f9d
From: "Hello Worldz" <sip:888@10.1.7.17>;tag=44c160be400f81ado0
To: <sip:1000@10.1.7.17>;tag=as1174f139
Call-ID: cf21c6d2-e12232a9@10.1.7.167
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="888",realm="asterisk",nonce="3f9e445f",uri="sip:1000@10.1.7.17",algorithm=MD5,response="dbb7339191ad45fc9fd9a732de1156a6"
Contact: "Hello Worldz" <sip:888@10.1.7.167:5060>
User-Agent: Sipura/SPA921-4.1.10(b)
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog 'cf21c6d2-e12232a9@10.1.7.167' Method: ACK
<--- SIP read from 10.1.7.167:5060 --->
INVITE sip:8500@10.1.7.17 SIP/2.0
Via: SIP/2.0/UDP 10.1.7.167:5060;branch=z9hG4bK-32e43c9e
From: "Hello Worldz" <sip:888@10.1.7.17>;tag=99baf5751aecae3eo0
To: <sip:8500@10.1.7.17>
Call-ID: 844a301-cd7dfe22@10.1.7.167
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "Hello Worldz" <sip:888@10.1.7.167:5060>
Expires: 240
User-Agent: Sipura/SPA921-4.1.10(b)
Content-Length: 389
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp
v=0
o=- 96944 96944 IN IP4 10.1.7.167
s=-
c=IN IP4 10.1.7.167
t=0 0
m=audio 16482 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/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:20
a=sendrecv
<------------->
--- (13 headers 18 lines) ---
== Using TOS bits 0
== Using CoS mark 0
Sending to 10.1.7.167 : 5060 (no NAT)
Using INVITE request as basis request - 844a301-cd7dfe22@10.1.7.167
No user '888' in SIP users list
Found peer '888' for '888' from 10.1.7.167:5060
<--- Reliably Transmitting (no NAT) to 10.1.7.167:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.7.167:5060;branch=z9hG4bK-32e43c9e;received=10.1.7.167
From: "Hello Worldz" <sip:888@10.1.7.17>;tag=99baf5751aecae3eo0
To: <sip:8500@10.1.7.17>;tag=as4dfa42cc
Call-ID: 844a301-cd7dfe22@10.1.7.167
CSeq: 101 INVITE
User-Agent: Asterisk PBX SVN-trunk-r63330
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="385e75ef"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '844a301-cd7dfe22@10.1.7.167' in 32000 ms (Method: INVITE)
<--- SIP read from 10.1.7.167:5060 --->
ACK sip:8500@10.1.7.17 SIP/2.0
Via: SIP/2.0/UDP 10.1.7.167:5060;branch=z9hG4bK-32e43c9e
From: "Hello Worldz" <sip:888@10.1.7.17>;tag=99baf5751aecae3eo0
To: <sip:8500@10.1.7.17>;tag=as4dfa42cc
Call-ID: 844a301-cd7dfe22@10.1.7.167
CSeq: 101 ACK
Max-Forwards: 70
Contact: "Hello Worldz" <sip:888@10.1.7.167:5060>
User-Agent: Sipura/SPA921-4.1.10(b)
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from 10.1.7.167:5060 --->
INVITE sip:8500@10.1.7.17 SIP/2.0
Via: SIP/2.0/UDP 10.1.7.167:5060;branch=z9hG4bK-afa0de03
From: "Hello Worldz" <sip:888@10.1.7.17>;tag=99baf5751aecae3eo0
To: <sip:8500@10.1.7.17>
Call-ID: 844a301-cd7dfe22@10.1.7.167
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="888",realm="asterisk",nonce="385e75ef",uri="sip:8500@10.1.7.17",algorithm=MD5,response="46bedbb6a6bcc4b0f35eb0b05903c07e"
Contact: "Hello Worldz" <sip:888@10.1.7.167:5060>
Expires: 240
User-Agent: Sipura/SPA921-4.1.10(b)
Content-Length: 389
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp
v=0
o=- 96944 96944 IN IP4 10.1.7.167
s=-
c=IN IP4 10.1.7.167
t=0 0
m=audio 16482 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/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:20
a=sendrecv
<------------->
--- (14 headers 18 lines) ---
Sending to 10.1.7.167 : 5060 (no NAT)
Using INVITE request as basis request - 844a301-cd7dfe22@10.1.7.167
No user '888' in SIP users list
Found peer '888' for '888' from 10.1.7.167:5060
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
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 10.1.7.167:16482
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 G729a for ID 18
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 - 0x4 (ulaw), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.1.7.167:16482
Looking for 8500 in users (domain 10.1.7.17)
<--- Reliably Transmitting (no NAT) to 10.1.7.167:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.1.7.167:5060;branch=z9hG4bK-afa0de03;received=10.1.7.167
From: "Hello Worldz" <sip:888@10.1.7.17>;tag=99baf5751aecae3eo0
To: <sip:8500@10.1.7.17>;tag=as4dfa42cc
Call-ID: 844a301-cd7dfe22@10.1.7.167
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-trunk-r63330
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '844a301-cd7dfe22@10.1.7.167' in 32000 ms (Method: INVITE)
<--- SIP read from 10.1.7.167:5060 --->
ACK sip:8500@10.1.7.17 SIP/2.0
Via: SIP/2.0/UDP 10.1.7.167:5060;branch=z9hG4bK-afa0de03
From: "Hello Worldz" <sip:888@10.1.7.17>;tag=99baf5751aecae3eo0
To: <sip:8500@10.1.7.17>;tag=as4dfa42cc
Call-ID: 844a301-cd7dfe22@10.1.7.167
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="888",realm="asterisk",nonce="385e75ef",uri="sip:8500@10.1.7.17",algorithm=MD5,response="c170d8c61e1df055e49e17b569465853"
Contact: "Hello Worldz" <sip:888@10.1.7.167:5060>
User-Agent: Sipura/SPA921-4.1.10(b)
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '844a301-cd7dfe22@10.1.7.167' Method: ACK