<--- SIP read from 172.24.8.130:50668 --->
INVITE sip:4802214606@172.24.8.36 SIP/2.0
Via: SIP/2.0/UDP 172.24.8.130:5060;branch=z9hG4bK7d98b4dd
From: "101" <sip:101@172.24.8.36>;tag=00115c430951000d17b0ca06-2e282a4f
To: <sip:4802214606@172.24.8.36>
Call-ID: 00115c43-0951000e-110946c5-2c7553b2@172.24.8.130
CSeq: 101 INVITE
User-Agent: CSCO/6
Contact: <sip:101@172.24.8.130:5060>
Expires: 180
Content-Type: application/sdp
Content-Length: 246
Accept: application/sdp
v=0
o=Cisco-SIPUA 21896 3724 IN IP4 172.24.8.130
s=SIP Call
c=IN IP4 172.24.8.130
t=0 0
m=audio 20698 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (12 headers 11 lines) ---
Sending to 172.24.8.130 : 5060 (no NAT)
Using INVITE request as basis request - 00115c43-0951000e-110946c5-2c7553b2@172.24.8.130
Found no matching peer or user for '172.24.8.130:50668'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 172.24.8.130:20698
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format G729 for ID 18
Found description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.24.8.130:20698
Looking for 4802214606 in default (domain 172.24.8.36)
localhost*CLI>
<--- Reliably Transmitting (no NAT) to 172.24.8.130:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.24.8.130:5060;branch=z9hG4bK7d98b4dd;received=172.24.8.130
From: "101" <sip:101@172.24.8.36>;tag=00115c430951000d17b0ca06-2e282a4f
To: <sip:4802214606@172.24.8.36>;tag=as461c86d0
Call-ID: 00115c43-0951000e-110946c5-2c7553b2@172.24.8.130
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '00115c43-0951000e-110946c5-2c7553b2@172.24.8.130' in 32000 ms (Method: INVITE)
localhost*CLI>
<--- SIP read from 172.24.8.130:50669 --->
ACK sip:4802214606@172.24.8.36 SIP/2.0
Via: SIP/2.0/UDP 172.24.8.130:5060;branch=z9hG4bK7d98b4dd
From: "101" <sip:101@172.24.8.36>;tag=00115c430951000d17b0ca06-2e282a4f
To: <sip:4802214606@172.24.8.36>;tag=as461c86d0
Call-ID: 00115c43-0951000e-110946c5-2c7553b2@172.24.8.130
CSeq: 101 ACK
Content-Length: 0