rendered paste body<--- SIP read from 10.4.20.184:5060 --->
INVITE sip:1000@10.4.20.111 SIP/2.0
Via: SIP/2.0/UDP 10.4.20.184:5060;rport;branch=z9hG4bK223040
To: <sip:1000@10.4.20.111>
From: "Joe S. Phigan" <sip:7101@10.4.20.111>;tag=7531
Call-ID: 1180028884-3040-TOPLAP@10.4.20.184
CSeq: 897 INVITE
Max-Forwards: 20
User-Agent: Uplink Skype2Sip 1.1
Contact: <sip:7101@10.4.20.184:5060>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 284
v=0
o=- 836899966 836899988 IN IP4 10.4.20.184
s=Uplink Skype2Sip
c=IN IP4 10.4.20.184
t=0 0
m=audio 10000 RTP/AVP 0 8 3 13 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (13 headers 14 lines) ---
Sending to 10.4.20.184 : 5060 (NAT)
Using INVITE request as basis request - 1180028884-3040-TOPLAP@10.4.20.184
<--- Reliably Transmitting (NAT) to 10.4.20.184:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.4.20.184:5060;branch=z9hG4bK223040;received=10.4.20.184;rport=5060
From: "Joe S. Phigan" <sip:7101@10.4.20.111>;tag=7531
To: <sip:1000@10.4.20.111>;tag=as79f6c802
Call-ID: 1180028884-3040-TOPLAP@10.4.20.184
CSeq: 897 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0163fef3"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1180028884-3040-TOPLAP@10.4.20.184' in 32000 ms (Method: INVITE)
Found user '7101'
Itchy*CLI>
<--- SIP read from 10.4.20.184:5060 --->
<------------->
--- (0 headers 0 lines) Nat keepalive ---
Itchy*CLI>
<--- SIP read from 10.4.20.184:5060 --->
ACK sip:1000@10.4.20.111 SIP/2.0
Via: SIP/2.0/UDP 10.4.20.184:5060;rport;branch=z9hG4bK223040
To: <sip:1000@10.4.20.111>;tag=as79f6c802
From: "Joe S. Phigan" <sip:7101@10.4.20.111>;tag=7531
Call-ID: 1180028884-3040-TOPLAP@10.4.20.184
CSeq: 897 ACK
Max-Forwards: 20
User-Agent: Uplink Skype2Sip 1.1
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Itchy*CLI>
<--- SIP read from 10.4.20.184:5060 --->
INVITE sip:1000@10.4.20.111 SIP/2.0
Via: SIP/2.0/UDP 10.4.20.184:5060;rport;branch=z9hG4bK233040
To: <sip:1000@10.4.20.111>
From: "Joe S. Phigan" <sip:7101@10.4.20.111>;tag=7531
Call-ID: 1180028884-3040-TOPLAP@10.4.20.184
CSeq: 898 INVITE
Max-Forwards: 20
User-Agent: Uplink Skype2Sip 1.1
Contact: <sip:7101@10.4.20.184:5060>
Proxy-Authorization: Digest username="7101",realm="asterisk",nonce="0163fef3",uri="sip:1000@10.4.20.111",response="fe2758259bfc973c20f55630aa51dcf8",opaque=""
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 284
v=0
o=- 836899966 836899988 IN IP4 10.4.20.184
s=Uplink Skype2Sip
c=IN IP4 10.4.20.184
t=0 0
m=audio 10000 RTP/AVP 0 8 3 13 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:13 CN/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (14 headers 14 lines) ---
Sending to 10.4.20.184 : 5060 (NAT)
Using INVITE request as basis request - 1180028884-3040-TOPLAP@10.4.20.184
Found user '7101'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 13
Found RTP audio format 101
Peer audio RTP is at port 10.4.20.184:10000
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format GSM for ID 3
Found description format CN for ID 13
Found description format telephone-event for ID 101
Capabilities: us - 0x3f0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|jpeg|png|h261|h263|h263p|h264), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.4.20.184:10000
Looking for 1000 in fromskype (domain 10.4.20.111)
list_route: hop: <sip:7101@10.4.20.184:5060>
<--- Transmitting (NAT) to 10.4.20.184:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.4.20.184:5060;branch=z9hG4bK233040;received=10.4.20.184;rport=5060
From: "Joe S. Phigan" <sip:7101@10.4.20.111>;tag=7531
To: <sip:1000@10.4.20.111>
Call-ID: 1180028884-3040-TOPLAP@10.4.20.184
CSeq: 898 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1000@10.4.20.111>
Content-Length: 0
<------------>
-- Executing [1000@fromskype:1] Dial("SIP/7101-081d70b0", "Zap/2&SIP/101&SIP/102&SIP/103") in new stack
[May 24 11:10:47] WARNING[13176]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown)
Really destroying SIP dialog '709d18be6b66c93a0874eb542a8d7f8e@127.0.1.1' Method: INVITE
[May 24 11:10:47] WARNING[13176]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
Really destroying SIP dialog '3052319774cbdbe91384b7494f849786@127.0.1.1' Method: INVITE
[May 24 11:10:47] WARNING[13176]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
Audio is at 10.4.20.111 port 10002
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.4.20.190:5060:
INVITE sip:103@10.4.20.190:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK04fde0c1;rport
From: "From Skype" <sip:7101@10.4.20.111>;tag=as4bfc4222
To: <sip:103@10.4.20.190:5060;user=phone>
Contact: <sip:7101@10.4.20.111>
Call-ID: 483533dc355c3c356e728cf63f9df4cb@10.4.20.111
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 24 May 2007 18:10:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 271
v=0
o=root 5910 5910 IN IP4 10.4.20.111
s=session
c=IN IP4 10.4.20.111
t=0 0
m=audio 10002 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv
---
-- Called 103
Itchy*CLI>
<--- SIP read from 10.4.20.190:5060 --->
SIP/2.0 100 Trying
Call-ID: 483533dc355c3c356e728cf63f9df4cb@10.4.20.111
From: "From Skype"<sip:7101@10.4.20.111>;tag=as4bfc4222
To: <sip:103@10.4.20.190:5060;user=phone>;tag=3c0f0-98612663
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK04fde0c1;rport
Contact: Cordless<sip:103@10.4.20.190:5060;user=phone>
Server: DTA SIP/0.12.35 DNLD/1.1.13
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Itchy*CLI>
<--- SIP read from 10.4.20.190:5060 --->
SIP/2.0 180 Ringing
Call-ID: 483533dc355c3c356e728cf63f9df4cb@10.4.20.111
From: "From Skype"<sip:7101@10.4.20.111>;tag=as4bfc4222
To: <sip:103@10.4.20.190:5060;user=phone>;tag=3c0f0-98612663
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK04fde0c1;rport
Contact: Cordless<sip:103@10.4.20.190:5060;user=phone>
Server: DTA SIP/0.12.35 DNLD/1.1.13
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
-- SIP/103-081db1e0 is ringing
<--- Transmitting (NAT) to 10.4.20.184:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.4.20.184:5060;branch=z9hG4bK233040;received=10.4.20.184;rport=5060
From: "Joe S. Phigan" <sip:7101@10.4.20.111>;tag=7531
To: <sip:1000@10.4.20.111>;tag=as6a0aa85e
Call-ID: 1180028884-3040-TOPLAP@10.4.20.184
CSeq: 898 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1000@10.4.20.111>
Content-Length: 0
<------------>
Itchy*CLI>
<--- SIP read from 10.4.20.190:5060 --->
H
<------------->
--- (1 headers 0 lines) ---
Itchy*CLI>
<--- SIP read from 10.4.20.190:5060 --->
SIP/2.0 200 OK
Call-ID: 483533dc355c3c356e728cf63f9df4cb@10.4.20.111
From: "From Skype"<sip:7101@10.4.20.111>;tag=as4bfc4222
To: <sip:103@10.4.20.190:5060;user=phone>;tag=3c0f0-98612663
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK04fde0c1;rport
Contact: Cordless<sip:103@10.4.20.190:5060;user=phone>
Supported: timer,100rel
Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS,PRACK
Server: DTA SIP/0.12.35 DNLD/1.1.13
Content-Type: application/sdp
Content-Length: 132
v=0
o=103 0 0 IN IP4 10.4.20.190
s=-
c=IN IP4 10.4.20.190
t=0 0
m=audio 8006 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
<------------->
--- (12 headers 7 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 10.4.20.190:8006
Found description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=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 10.4.20.190:8006
list_route: hop: <sip:103@10.4.20.190:5060;user=phone>
set_destination: Parsing <sip:103@10.4.20.190:5060;user=phone> for address/port to send to
set_destination: set destination to 10.4.20.190, port 5060
Transmitting (NAT) to 10.4.20.190:5060:
ACK sip:103@10.4.20.190:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK1744bff9;rport
From: "From Skype" <sip:7101@10.4.20.111>;tag=as4bfc4222
To: <sip:103@10.4.20.190:5060;user=phone>;tag=3c0f0-98612663
Contact: <sip:7101@10.4.20.111>
Call-ID: 483533dc355c3c356e728cf63f9df4cb@10.4.20.111
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/103-081db1e0 answered SIP/7101-081d70b0
Audio is at 10.4.20.111 port 10014
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 10.4.20.184:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.4.20.184:5060;branch=z9hG4bK233040;received=10.4.20.184;rport=5060
From: "Joe S. Phigan" <sip:7101@10.4.20.111>;tag=7531
To: <sip:1000@10.4.20.111>;tag=as6a0aa85e
Call-ID: 1180028884-3040-TOPLAP@10.4.20.184
CSeq: 898 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1000@10.4.20.111>
Content-Type: application/sdp
Content-Length: 271
v=0
o=root 5910 5910 IN IP4 10.4.20.111
s=session
c=IN IP4 10.4.20.111
t=0 0
m=audio 10014 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
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=sendrecv
<------------>
-- Packet2Packet bridging SIP/7101-081d70b0 and SIP/103-081db1e0
Itchy*CLI>
<--- SIP read from 10.4.20.184:5060 --->
ACK sip:1000@10.4.20.111 SIP/2.0
Via: SIP/2.0/UDP 10.4.20.184:5060;rport;branch=z9hG4bK243040
To: <sip:1000@10.4.20.111>;tag=as6a0aa85e
From: "Joe S. Phigan" <sip:7101@10.4.20.111>;tag=7531
Call-ID: 1180028884-3040-TOPLAP@10.4.20.184
CSeq: 898 ACK
Max-Forwards: 20
User-Agent: Uplink Skype2Sip 1.1
Proxy-Authorization: Digest username="7101",realm="asterisk",nonce="0163fef3",uri="sip:1000@10.4.20.111",response="fe2758259bfc973c20f55630aa51dcf8",opaque=""
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '1180028882-3040-TOPLAP@10.4.20.184' Method: ACK
Really destroying SIP dialog '3ad373b10b6946c13dea557f3bbd52b8@127.0.1.1' Method: REGISTER
Itchy*CLI>
<--- SIP read from 10.4.20.190:5060 --->
H
<------------->
--- (1 headers 0 lines) ---
Itchy*CLI>
<--- SIP read from 10.4.20.184:5060 --->
BYE sip:1000@10.4.20.111 SIP/2.0
Via: SIP/2.0/UDP 10.4.20.184:5060;rport;branch=z9hG4bK253040
To: <sip:1000@10.4.20.111>;tag=as6a0aa85e
From: "Joe S. Phigan" <sip:7101@10.4.20.111>;tag=7531
Call-ID: 1180028884-3040-TOPLAP@10.4.20.184
CSeq: 899 BYE
Max-Forwards: 20
User-Agent: Uplink Skype2Sip 1.1
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Sending to 10.4.20.184 : 5060 (NAT)
<--- Transmitting (NAT) to 10.4.20.184:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.4.20.184:5060;branch=z9hG4bK253040;received=10.4.20.184;rport=5060
From: "Joe S. Phigan" <sip:7101@10.4.20.111>;tag=7531
To: <sip:1000@10.4.20.111>;tag=as6a0aa85e
Call-ID: 1180028884-3040-TOPLAP@10.4.20.184
CSeq: 899 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1000@10.4.20.111>
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '483533dc355c3c356e728cf63f9df4cb@10.4.20.111' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:103@10.4.20.190:5060;user=phone> for address/port to send to
set_destination: set destination to 10.4.20.190, port 5060
Reliably Transmitting (NAT) to 10.4.20.190:5060:
BYE sip:103@10.4.20.190:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK557cc0fd;rport
From: "From Skype" <sip:7101@10.4.20.111>;tag=as4bfc4222
To: <sip:103@10.4.20.190:5060;user=phone>;tag=3c0f0-98612663
Contact: <sip:7101@10.4.20.111>
all-ID: 483533dc355c3c356e728cf63f9df4cb@10.4.20.111
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
== Spawn extension (fromskype, 1000, 1) exited non-zero on 'SIP/7101-081d70b0'
Itchy*CLI>
<--- SIP read from 10.4.20.190:5060 --->
SIP/2.0 200 OK
Call-ID: 483533dc355c3c356e728cf63f9df4cb@10.4.20.111
From: "From Skype"<sip:7101@10.4.20.111>;tag=as4bfc4222
To: <sip:103@10.4.20.190:5060;user=phone>;tag=3c0f0-98612663
CSeq: 103 BYE
Via: SIP/2.0/UDP 10.4.20.111:5060;branch=z9hG4bK557cc0fd;rport
Contact: Cordless<sip:103@10.4.20.190:5060;user=phone>
Server: DTA SIP/0.12.35 DNLD/1.1.13
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '483533dc355c3c356e728cf63f9df4cb@10.4.20.111' Method: INVITE
Really destroying SIP dialog '1180028884-3040-TOPLAP@10.4.20.184' Method: BYE