SIP Debugging Enabled
*** Sip read: 
INVITE sip:814@216.251.128.22;user=phone SIP/2.0
Via: SIP/2.0/UDP 216.251.134.7:5060
From: sip:greg@216.251.128.22;tag=2801500346
To: <sip:814@216.251.128.22;user=phone>
Call-ID: 799160957@216.251.134.7
CSeq: 1 INVITE
Contact: <sip:greg@216.251.134.7:5060;transport=udp>
User-Agent: Cisco ATA 186  v2.16.2 ata18x (030909a)
Expires: 300
Content-Length: 255
Content-Type: application/sdp

v=0
o=greg 4383835 4383835 IN IP4 216.251.134.7
s=ATA186 Call
c=IN IP4 216.251.134.7
t=0 0
=audio 16384 RTP/AVP 18 8 0 101
a=rtpmap:18 G729/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

 -- parsed 11 headers, 11 lines
 -- Using latest request as basis request
 -- Sending to 216.251.134.7 : 5060
 -- Found audio format UNKN
 -- Found audio format ALAW
 -- Found audio format UNKN
 -- Found audio format UNKN
 -- Found description format G729
 -- Found description format PCMA
 -- Found description format PCMU
 -- Found description format telephone-event
 -- Capabilities: us - 260, them - 268/0, combined - 260
 -- Non-codec capabilities: us - 1, them - 1, combined - 1
 -- Looking for 814 in incoming
list_route: hop: <sip:greg@216.251.134.7:5060;transport=udp>
*** Transmitting Response to 216.251.134.7:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 216.251.134.7:5060
From: sip:greg@216.251.128.22;tag=2801500346
To: <sip:814@216.251.128.22;user=phone>;tag=as20a622c4
Call-ID: 799160957@216.251.134.7
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:814@216.251.128.22>
Content-Length: 0


 -- We're at 216.251.128.22 port 11568
 -- Answering with preferred capability 4
 -- Answering with preferred capability 256
 -- Answering with non-codec capability 1
*** Reliably Transmitting Response to 216.251.134.7:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.251.134.7:5060
From: sip:greg@216.251.128.22;tag=2801500346
To: <sip:814@216.251.128.22;user=phone>;tag=as20a622c4
Call-ID: 799160957@216.251.134.7
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:814@216.251.128.22>
Content-Type: application/sdp
Content-Length: 219

v=0
o=root 11545 11545 IN IP4 216.251.128.22
s=session
c=IN IP4 216.251.128.22
t=0 0
m=audio 11568 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

*** Sip read: 
ACK sip:814@216.251.128.22 SIP/2.0
Via: SIP/2.0/UDP 216.251.134.7:5060
From: sip:greg@216.251.128.22;tag=2801500346
To: <sip:814@216.251.128.22;user=phone>;tag=as20a622c4
Call-ID: 799160957@216.251.134.7
CSeq: 1 ACK
User-Agent: Cisco ATA 186  v2.16.2 ata18x (030909a)
Content-Length: 0


 -- parsed 8 headers, 0 lines
*** Sip read: 


 -- parsed 0 headers, 0 lines
 -- parsed 10 headers, 0 lines
*** Reliably Transmitting Request to 216.251.130.76:5060
OPTIONS sip:216.251.130.76 SIP/2.0
Via: SIP/2.0/UDP 216.251.128.22:5060;branch=z9hG4bK576d28b1
From: "asterisk" <sip:asterisk@216.251.128.22>;tag=as47e73832
To: <sip:216.251.130.76>
Contact: <sip:asterisk@216.251.128.22>
Call-ID: 62ee31043dfa737a686654e26eed8363@216.251.128.22
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0


*** Sip read: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.251.128.22:5060;branch=z9hG4bK576d28b1
From: "asterisk" <sip:asterisk@216.251.128.22>;tag=as47e73832
To: <sip:216.251.130.76>;tag=2728664720
Call-ID: 62ee31043dfa737a686654e26eed8363@216.251.128.22
CSeq: 102 OPTIONS
Server: Cisco ATA 186  v2.15 ata18x (030122a)
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REGISTER
Content-Length: 261
Content-Type: application/sdp

v=0
o=chris0 15071714 15071714 IN IP4 216.251.130.76
s=ATA186 Call
c=IN IP4 216.251.130.76
t=0 0
m=audio 16384 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

 -- parsed 10 headers, 11 lines
 -- parsed 10 headers, 0 lines