Code:
[Jun 9 16:42:04]
<--- SIP read from UDP:172.16.0.1:61402 --->
INVITE sip:3072@172.16.128.1:5060 SIP/2.0
Date: Thu, 09 Jun 2011 14:41:11 GMT
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
From: "9234" <sip:9234@172.16.0.1>;tag=E1EE50-198B
Allow-Events: telephone-event
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Remote-Party-ID: "9234" <sip:9234@172.16.0.1>;party=calling;screen=no;privacy=off
Cisco-Guid: 1493594418-2447184352-2180040113-699679628
Timestamp: 1307630471
Content-Length: 234
User-Agent: Cisco-SIPGateway/IOS-12.x
To: <sip:3072@172.16.128.1>
Contact: <sip:9234@172.16.0.1:5060>
Expires: 180
Content-Disposition: session;handling=required
Content-Type: application/sdp
Call-ID: 5A9EC57E-91DD11E0-81F5C5B1-29B4438C@172.16.0.1
Via: SIP/2.0/UDP 172.16.0.1:5060;branch=z9hG4bK121B80
CSeq: 101 INVITE
Max-Forwards: 70
v=0
o=CiscoSystemsSIP-GW-UserAgent 1707 5598 IN IP4 172.16.0.1
s=SIP Call
c=IN IP4 172.16.0.1
t=0 0
m=audio 19264 RTP/AVP 0 100
c=IN IP4 172.16.0.1
a=rtpmap:0 PCMU/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=ptime:20
<------------->
[Jun 9 16:42:04] --- (21 headers 11 lines) ---
[Jun 9 16:42:04] Sending to 172.16.0.1 : 5060 (no NAT)
[Jun 9 16:42:04] Using INVITE request as basis request - 5A9EC57E-91DD11E0-81F5C5B1-29B4438C@172.16.0.1
[Jun 9 16:42:04] No matching peer for '9234' from '172.16.0.1:61402'
[Jun 9 16:42:04] Found RTP audio format 0
[Jun 9 16:42:04] Found RTP audio format 100
[Jun 9 16:42:04] Found audio description format PCMU for ID 0
[Jun 9 16:42:04] Found audio description format X-NSE for ID 100
[Jun 9 16:42:04] Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
[Jun 9 16:42:04] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Jun 9 16:42:04] Peer audio RTP is at port 172.16.0.1:19264
[Jun 9 16:42:04] Looking for 3072 in default (domain 172.16.128.1)
[Jun 9 16:42:04]
<--- Reliably Transmitting (no NAT) to 172.16.0.1:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.16.0.1:5060;branch=z9hG4bK121B80;received=172.16.0.1
From: "9234" <sip:9234@172.16.0.1>;tag=E1EE50-198B
To: <sip:3072@172.16.128.1>;tag=as39d4a141
Call-ID: 5A9EC57E-91DD11E0-81F5C5B1-29B4438C@172.16.0.1
CSeq: 101 INVITE
Server: Asterisk PBX 1.6.2.7-1ubuntu1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
asterisk*CLI>
<------------>
[Jun 9 16:42:04] NOTICE[5235]: chan_sip.c:20063 handle_request_invite: Call from '' to extension '3072' rejected because extension not found.
[Jun 9 16:42:04] Scheduling destruction of SIP dialog '5A9EC57E-91DD11E0-81F5C5B1-29B4438C@172.16.0.1' in 32000 ms (Method: INVITE)
[Jun 9 16:42:04]
<--- SIP read from UDP:172.16.0.1:61402 --->
ACK sip:3072@172.16.128.1:5060 SIP/2.0
Date: Thu, 09 Jun 2011 14:41:11 GMT
From: "9234" <sip:9234@172.16.0.1>;tag=E1EE50-198B
Allow-Events: telephone-event
Content-Length: 0
To: <sip:3072@172.16.128.1>;tag=as39d4a141
Call-ID: 5A9EC57E-91DD11E0-81F5C5B1-29B4438C@172.16.0.1
Via: SIP/2.0/UDP 172.16.0.1:5060;branch=z9hG4bK121B80
CSeq: 101 ACK
Max-Forwards: 70
<------------->
[Jun 9 16:42:04] --- (10 headers 0 lines) ---
[Jun 9 16:42:04] Really destroying SIP dialog '5A9EC57E-91DD11E0-81F5C5B1-29B4438C@172.16.0.1' Method: ACK
[Jun 9 16:42:15] Reliably Transmitting (NAT) to 172.16.0.1:5060:
OPTIONS sip:172.16.0.1 SIP/2.0
Via: SIP/2.0/UDP 172.16.128.1:5060;branch=z9hG4bK2337bddc;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.16.128.1>;tag=as30adae72
To: <sip:172.16.0.1>
Contact: <sip:asterisk@172.16.128.1>
Call-ID: 220b59343dfbf722703fc39c73654811@172.16.128.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.7-1ubuntu1.1
Date: Thu, 09 Jun 2011 14:42:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
[Jun 9 16:42:15]
<--- SIP read from UDP:172.16.0.1:5060 --->
SIP/2.0 200 OK
Date: Thu, 09 Jun 2011 14:41:22 GMT
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
From: "asterisk" <sip:asterisk@172.16.128.1>;tag=as30adae72
Allow-Events: telephone-event
Supported: 100rel,resource-priority,replaces,sdp-anat
Content-Length: 163
To: <sip:172.16.0.1>;tag=E21AE0-20DA
Content-Type: application/sdp
Call-ID: 220b59343dfbf722703fc39c73654811@172.16.128.1
Accept: application/sdp
Via: SIP/2.0/UDP 172.16.128.1:5060;branch=z9hG4bK2337bddc;rport
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
v=0
o=CiscoSystemsSIP-GW-UserAgent 4727 5364 IN IP4 172.16.0.1
s=SIP Call
c=IN IP4 172.16.0.1
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 172.16.0.1
<------------->
[Jun 9 16:42:15] --- (14 headers 7 lines) ---
[Jun 9 16:42:15] Really destroying SIP dialog '220b59343dfbf722703fc39c73654811@172.16.128.1' Method: OPTIONS