rendered paste body<-- SIP read from 192.168.17.1:54567:
INVITE sip:4601@192.168.17.20:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.17.1:5060;branch=z9hG4bK7E2052
Remote-Party-ID: "Riela casa" <sip:603@192.168.17.1>;party=calling;screen=no;privacy=off
From: "Riela casa" <sip:5302325@192.168.17.20>;tag=471177C-1844
To: <sip:4601@192.168.17.20>
Date: Tue, 31 Oct 2006 18:28:54 GMT
Call-ID: 7F7A4506-684411DB-8725DFF0-3FBAA4A@192.168.17.1
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 2130319448-1749291483-2267078640-66824778
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1162319334
Contact: <sip:5302325@192.168.17.1:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 283
v=0
o=CiscoSystemsSIP-GW-UserAgent 4599 8370 IN IP4 192.168.17.1
s=SIP Call
c=IN IP4 192.168.17.1
t=0 0
m=audio 17162 RTP/AVP 8 0 101 19
c=IN IP4 192.168.17.1
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
--- (21 headers 12 lines)---
Using INVITE request as basis request - 7F7A4506-684411DB-8725DFF0-3FBAA4A@192.168.17.1
Sending to 192.168.17.1 : 5060 (non-NAT)
Found no matching peer or user for '192.168.17.1:54567'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found RTP audio format 19
Peer audio RTP is at port 192.168.17.1:17162
Found description format PCMA
Found description format PCMU
Found description format telephone-event
Found description format CN
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event)
Looking for 4601 in from-sip-external (domain 192.168.17.20)
list_route: hop: <sip:5302325@192.168.17.1:5060>
Transmitting (no NAT) to 192.168.17.1:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.17.1:5060;branch=z9hG4bK7E2052;received=192.168.17.1
From: "Riela casa" <sip:5302325@192.168.17.20>;tag=471177C-1844
To: <sip:4601@192.168.17.20>
Call-ID: 7F7A4506-684411DB-8725DFF0-3FBAA4A@192.168.17.1
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:4601@192.168.17.20>
Content-Length: 0
---
We're at 192.168.17.20 port 12578
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.17.1:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.17.1:5060;branch=z9hG4bK7E2052;received=192.168.17.1
From: "Riela casa" <sip:5302325@192.168.17.20>;tag=471177C-1844
To: <sip:4601@192.168.17.20>;tag=as34dc1f65
Call-ID: 7F7A4506-684411DB-8725DFF0-3FBAA4A@192.168.17.1
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:4601@192.168.17.20>
Content-Type: application/sdp
Content-Length: 242
v=0
o=root 14986 14986 IN IP4 192.168.17.20
s=session
c=IN IP4 192.168.17.20
t=0 0
m=audio 12578 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
asterisk1*CLI>
<-- SIP read from 192.168.17.1:54567:
ACK sip:4601@192.168.17.20:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.17.1:5060;branch=z9hG4bK7F4B8
From: "Riela casa" <sip:5302325@192.168.17.20>;tag=471177C-1844
To: <sip:4601@192.168.17.20>;tag=as34dc1f65
Date: Tue, 31 Oct 2006 18:28:54 GMT
Call-ID: 7F7A4506-684411DB-8725DFF0-3FBAA4A@192.168.17.1
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
--- (10 headers 0 lines)---
asterisk1*CLI>
<-- SIP read from 192.168.17.1:54567:
BYE sip:4601@192.168.17.20:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.17.1:5060;branch=z9hG4bK80A4A
From: "Riela casa" <sip:5302325@192.168.17.20>;tag=471177C-1844
To: <sip:4601@192.168.17.20>;tag=as34dc1f65
Date: Tue, 31 Oct 2006 18:28:54 GMT
Call-ID: 7F7A4506-684411DB-8725DFF0-3FBAA4A@192.168.17.1
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1162319343
CSeq: 102 BYE
Reason: Q.850;cause=16
Content-Length: 0
--- (12 headers 0 lines)---
Sending to 192.168.17.1 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.17.1:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.17.1:5060;branch=z9hG4bK80A4A;received=192.168.17.1
From: "Riela casa" <sip:5302325@192.168.17.20>;tag=471177C-1844
To: <sip:4601@192.168.17.20>;tag=as34dc1f65
Call-ID: 7F7A4506-684411DB-8725DFF0-3FBAA4A@192.168.17.1
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:4601@192.168.17.20>
Content-Length: 0
---
Destroying call '7F7A4506-684411DB-8725DFF0-3FBAA4A@192.168.17.1'
asterisk1*CLI> sip no debug
SIP Debugging Disabled