All pastes #15625 Raw Edit

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#15625 ·published 2005-06-23 09:24 UTC
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INVITE sip:7001@192.168.0.200;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.201:5060
From: <sip:2299@192.168.0.200;user=phone>;tag=545885975
To: <sip:7001@192.168.0.200;user=phone>
Call-ID: 1683847546@192.168.0.201
CSeq: 1 INVITE
Contact: <sip:2299@192.168.0.201:5060;user=phone;transport=udp>
User-Agent: Cisco ATA 188  v3.1.1 atasip (040629A)
Expires: 300
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Supported: replaces
Content-Length: 249
Content-Type: application/sdp
o=2299 11146 11146 IN IP4 192.168.0.201
s=ATA186 Call
c=IN IP4 192.168.0.201
t=0 0
m=audio 16384 RTP/AVP 0 4 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
13 headers, 11 lines
Using latest request as basis request
Sending to 192.168.0.201 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.201:16384
Found description format PCMU
Found description format G723
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0xd(G723|ULAW|ALAW)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
Found user '2299'
Looking for 7001 in default
list_route: hop: <sip:2299@192.168.0.201:5060;user=phone;transport=udp>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.201:5060
From: <sip:2299@192.168.0.200;user=phone>;tag=545885975
To: <sip:7001@192.168.0.200;user=phone>;tag=as4728682c
Call-ID: 1683847546@192.168.0.201
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:7001@192.168.0.200>
Content-Length: 0
 to 192.168.0.201:5060
We're at 192.168.0.200 port 17084
Video is at 192.168.0.200 port 16692
Answering/Requesting with root capability 8
Answering with capability 0x2(GSM)
Answering with capability 0x4(ULAW)
Answering with capability 0x80000(H263)
Answering with non-codec capability 0x1(G723)
12 headers, 14 lines
Reliably Transmitting:
INVITE sip:7001@192.168.0.206 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK0ce634f0
From: "ATA Line 1" <sip:2299@192.168.0.200>;tag=as5539761f
To: <sip:7001@192.168.0.206>
Contact: <sip:2299@192.168.0.200>
Call-ID: 3722563f6b547b605d2b9d6f5f1ad756@192.168.0.200
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Mon, 16 Aug 2004 20:06:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 315
o=root 25260 25260 IN IP4 192.168.0.200
s=session
c=IN IP4 192.168.0.200
t=0 0
m=audio 17084 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
m=video 16692 RTP/AVP 34
a=rtpmap:34 H263/90000
 (no NAT) to 192.168.0.206:5060
[Kasterisk*CLI>
Transmitting (no NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.201:5060
From: <sip:2299@192.168.0.200;user=phone>;tag=545885975
To: <sip:7001@192.168.0.200;user=phone>;tag=as4728682c
Call-ID: 1683847546@192.168.0.201
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:7001@192.168.0.200>
Content-Length: 0
 to 192.168.0.201:5060
[Kasterisk*CLI>
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK0ce634f0
From: "ATA Line 1" <sip:2299@192.168.0.200>;tag=as5539761f
To: <sip:7001@192.168.0.206>;tag=464E90-1FE
Date: Fri, 01 Mar 2002 01:16:47 GMT
Call-ID: 3722563f6b547b605d2b9d6f5f1ad756@192.168.0.200
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
10 headers, 0 lines
[Kasterisk*CLI>
Sip read:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK0ce634f0
From: "ATA Line 1" <sip:2299@192.168.0.200>;tag=as5539761f
To: <sip:7001@192.168.0.206>;tag=464E90-1FE
Date: Fri, 01 Mar 2002 01:16:47 GMT
Call-ID: 3722563f6b547b605d2b9d6f5f1ad756@192.168.0.200
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: UPDATE
Allow-Events: telephone-event
Contact: <sip:7001@192.168.0.206:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 281
o=CiscoSystemsSIP-GW-UserAgent 6900 6347 IN IP4 192.168.0.206
s=SIP Call
c=IN IP4 192.168.0.206
t=0 0
m=audio 18900 RTP/AVP 8 101
c=IN IP4 192.168.0.206
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 0 RTP/AVP
c=IN IP4 192.168.0.206
14 headers, 12 lines
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.206:18900
Found description format PCMA
Found description format telephone-event
[Kasterisk*CLI>
Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x8(ALAW)/video=0x0(EMPTY), combined - 0x8(ALAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
[Kasterisk*CLI>
We're at 192.168.0.200 port 12878
Video is at 192.168.0.200 port 18752
Answering with capability 0x8(ALAW)
Answering with non-codec capability 0x1(G723)
Transmitting (no NAT):
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.201:5060
From: <sip:2299@192.168.0.200;user=phone>;tag=545885975
To: <sip:7001@192.168.0.200;user=phone>;tag=as4728682c
Call-ID: 1683847546@192.168.0.201
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:7001@192.168.0.200>
Content-Type: application/sdp
Content-Length: 218
o=root 25260 25260 IN IP4 192.168.0.200
s=session
c=IN IP4 192.168.0.200
t=0 0
m=audio 12878 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 to 192.168.0.201:5060
[Kasterisk*CLI>
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK0ce634f0
From: "ATA Line 1" <sip:2299@192.168.0.200>;tag=as5539761f
To: <sip:7001@192.168.0.206>;tag=464E90-1FE
Date: Fri, 01 Mar 2002 01:16:47 GMT
Call-ID: 3722563f6b547b605d2b9d6f5f1ad756@192.168.0.200
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <sip:7001@192.168.0.206:5060>
Content-Type: application/sdp
Content-Length: 281
o=CiscoSystemsSIP-GW-UserAgent 6900 6347 IN IP4 192.168.0.206
s=SIP Call
c=IN IP4 192.168.0.206
t=0 0
m=audio 18900 RTP/AVP 8 101
c=IN IP4 192.168.0.206
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 0 RTP/AVP
c=IN IP4 192.168.0.206
13 headers, 12 lines
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.206:18900
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x8(ALAW)/video=0x0(EMPTY), combined - 0x8(ALAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
list_route: hop: <sip:7001@192.168.0.206:5060>
set_destination: Parsing <sip:7001@192.168.0.206:5060> for address/port to send to
set_destination: set destination to 192.168.0.206, port 5060
Transmitting:
ACK sip:7001@192.168.0.206 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK4799a5d2
From: "ATA Line 1" <sip:2299@192.168.0.200>;tag=as5539761f
To: <sip:7001@192.168.0.206>;tag=464E90-1FE
Contact: <sip:2299@192.168.0.200>
Call-ID: 3722563f6b547b605d2b9d6f5f1ad756@192.168.0.200
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
 (no NAT) to 192.168.0.206:5060
We're at 192.168.0.200 port 12878
Video is at 192.168.0.200 port 18752
Answering with capability 0x8(ALAW)
Answering with non-codec capability 0x1(G723)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
[Kasterisk*CLI>
ia: SIP/2.0/UDP 192.168.0.201:5060
From: <sip:2299@192.168.0.200;user=phone>;tag=545885975
To: <sip:7001@192.168.0.200;user=phone>;tag=as4728682c
Call-ID: 1683847546@192.168.0.201
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:7001@192.168.0.200>
Content-Type: application/sdp
Content-Length: 218
o=root 25260 25261 IN IP4 192.168.0.200
s=session
c=IN IP4 192.168.0.200
t=0 0
m=audio 12878 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 to 192.168.0.201:5060
[Kasterisk*CLI>
Sip read:
ACK sip:7001@192.168.0.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.201:5060
From: <sip:2299@192.168.0.200;user=phone>;tag=545885975
To: <sip:7001@192.168.0.200;user=phone>;tag=as4728682c
Call-ID: 1683847546@192.168.0.201
CSeq: 1 ACK
User-Agent: Cisco ATA 188  v3.1.1 atasip (040629A)
Content-Length: 0
8 headers, 0 lines
[Kasterisk*CLI>
Sip read:
BYE sip:2299@192.168.0.200:5060 SIP/2.0
Via: SIP/2.0/UDP  192.168.0.206:5060;branch=z9hG4bK1934
From: <sip:7001@192.168.0.206>;tag=464E90-1FE
To: "ATA Line 1" <sip:2299@192.168.0.200>;tag=as5539761f
Date: Fri, 01 Mar 2002 01:16:48 GMT
Call-ID: 3722563f6b547b605d2b9d6f5f1ad756@192.168.0.200
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1014945412
CSeq: 101 BYE
Reason: Q.850;cause=16
Content-Length: 0
12 headers, 0 lines
Sending to 192.168.0.206 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.206:5060;branch=z9hG4bK1934
From: <sip:7001@192.168.0.206>;tag=464E90-1FE
To: "ATA Line 1" <sip:2299@192.168.0.200>;tag=as5539761f
Call-ID: 3722563f6b547b605d2b9d6f5f1ad756@192.168.0.200
CSeq: 101 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2299@192.168.0.200>
Content-Length: 0
 to 192.168.0.206:5060
[Kasterisk*CLI>
set_destination: Parsing <sip:2299@192.168.0.201:5060;user=phone;transport=udp> for address/port to send to
set_destination: set destination to 192.168.0.201, port 5060
Reliably Transmitting:
BYE sip:2299@192.168.0.201:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK1445925b;rport
From: <sip:7001@192.168.0.200;user=phone>;tag=as4728682c
To: <sip:2299@192.168.0.200;user=phone>;tag=545885975
Contact: <sip:7001@192.168.0.200>
Call-ID: 1683847546@192.168.0.201
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
 (no NAT) to 192.168.0.201:5060
[Kasterisk*CLI>
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.200:5060;branch=z9hG4bK1445925b;rport
From: <sip:7001@192.168.0.200;user=phone>;tag=as4728682c
To: <sip:2299@192.168.0.200;user=phone>;tag=545885975
Call-ID: 1683847546@192.168.0.201
CSeq: 102 BYE
Server: Cisco ATA 188  v3.1.1 atasip (040629A)
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Supported: replaces
Content-Length: 0
10 headers, 0 lines
Message is BYE
Destroying call '3722563f6b547b605d2b9d6f5f1ad756@192.168.0.200'
Destroying call '1683847546@192.168.0.201'
[Kasterisk*CLI>