rendered paste body<------------->
--- (0 headers 0 lines) Nat keepalive ---
Really destroying SIP dialog '5d955eda041e2792479cb9c9192723b6@rcps.onvoip.net' Method: REGISTER
<--- SIP read from 204.11.119.45:33382 --->
INVITE sip:7001@64.81.140.205 SIP/2.0
Via:SIP/2.0/UDP 204.11.119.45;branch=z9hG4bK-BroadWorks.204.11.119.45-64.81.140.205V5060-0-824191607-260617551-1174174455021-
From:"Cell Phone IL"<sip:7733194612@204.11.119.45;user=phone>;tag=260617551-1174174455021-
To:"Larry Butterfield Elvin Yonan Tech Contact"<sip:7737285106@rcps.onvoip.net;user=phone>
Call-ID:BW173415021170307-1152202556@204.11.119.45
CSeq:824191607 INVITE
Contact:<sip:204.11.119.45:5060>
Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY
Supported:100rel
Accept:multipart/mixed,application/sdp
Max-Forwards:10
Content-Type:application/sdp
Content-Length:107
v=0
o=BroadWorks 765727 1 IN IP4 4.68.248.2
s=-
c=IN IP4 4.68.248.2
t=0 0
m=audio 60862 RTP/AVP 0 18
<------------->
--- (13 headers 6 lines) ---
Sending to 204.11.119.45 : 5060 (no NAT)
Using INVITE request as basis request - BW173415021170307-1152202556@204.11.119.45
Found peer '7001'
Found RTP audio format 0
Found RTP audio format 18
Peer audio RTP is at port 4.68.248.2:60862
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 4.68.248.2:60862
Looking for 7001 in incomingcalls (domain 64.81.140.205)
<--- Reliably Transmitting (no NAT) to 204.11.119.45:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 204.11.119.45;branch=z9hG4bK-BroadWorks.204.11.119.45-64.81.140.205V5060-0-824191607-260617551-1174174455021-;received=204.11.119.45
From: "Cell Phone IL"<sip:7733194612@204.11.119.45;user=phone>;tag=260617551-1174174455021-
To: "Larry Butterfield Elvin Yonan Tech Contact"<sip:7737285106@rcps.onvoip.net;user=phone>;tag=as7e338d0c
Call-ID: BW173415021170307-1152202556@204.11.119.45
CSeq: 824191607 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'BW173415021170307-1152202556@204.11.119.45' in 32000 ms (Method: INVITE)
<--- SIP read from 204.11.119.45:33382 --->
ACK sip:7001@64.81.140.205 SIP/2.0
Via:SIP/2.0/UDP 204.11.119.45;branch=z9hG4bK-BroadWorks.204.11.119.45-64.81.140.205V5060-0-824191607-260617551-1174174455021-
From:"Cell Phone IL"<sip:7733194612@204.11.119.45;user=phone>;tag=260617551-1174174455021-
To:"Larry Butterfield Elvin Yonan Tech Contact"<sip:7737285106@rcps.onvoip.net;user=phone>;tag=as7e338d0c
Call-ID:BW173415021170307-1152202556@204.11.119.45
CSeq:824191607 ACK
Max-Forwards:10
Content-Length:0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'BW173415021170307-1152202556@204.11.119.45' Method: ACK
<--- SIP read from 10.1.1.50:5060 --->
<------------->
--- (0 headers 0 lines) Nat keepalive ---
<--- SIP read from 10.1.1.50:5060 --->
<------------->
--- (0 headers 0 lines) Nat keepalive ---
<--- SIP read from 10.1.1.50:5060 --->
<------------->
--- (0 headers 0 lines) Nat keepalive ---
[Mar 17 19:59:08] NOTICE[20346]: chan_sip.c:7055 sip_reregister: -- Re-registration for 7737285106@rcps.onvoip.net