rendered paste body*CLI>
<-- SIP read from 69.1.236.33:5060:
INVITE sip:s@12.206.2.116 SIP/2.0
Record-Route: <sip:+12179034857@69.1.236.33;ftag=VPSF50603522629637;lr>
Via: SIP/2.0/UDP 69.1.236.33;branch=z9hG4bKec6.39b501d4.1
Via: SIP/2.0/UDP 209.247.16.5:5060;branch=z9hG4bK50603522629637-1157004452658
From: "PCS Phone IL" <sip:+12177665959@209.247.16.5>;tag=VPSF50603522629637
To: <sip:+12179034857@69.1.236.33:5060>
Call-ID: DALMGC0320060930202747006680@209.244.62.68
CSeq: 1 INVITE
Contact: <sip:+12177665959@209.247.16.5:5060;transport=udp>
Max-Forwards: 68
Content-Type: application/sdp
Content-Length: 115
Remote-Party-ID: "PCS Phone IL" <sip:+12177665959@209.247.16.5>;party=calling;screen=yes;privacy=off
v=0
o=- 1159648067 1159648068 IN IP4 63.215.29.33
s=-
c=IN IP4 66.128.8.234
t=0 0
m=audio 52890 RTP/AVP 0 18
--- (13 headers 6 lines)---
Using INVITE request as basis request - DALMGC0320060930202747006680@209.244.62.68
Sending to 69.1.236.33 : 5060 (non-NAT)
Found peer 'sipmedia'
Found RTP audio format 0
Found RTP audio format 18
Peer audio RTP is at port 66.128.8.234:52890
Capabilities: us - 0x4 (ulaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Looking for s in sipurain (domain 12.206.2.116)
list_route: hop: <sip:+12179034857@69.1.236.33;ftag=VPSF50603522629637;lr>
Transmitting (no NAT) to 69.1.236.33:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 69.1.236.33;branch=z9hG4bKec6.39b501d4.1;received=69.1.236.33
Via: SIP/2.0/UDP 209.247.16.5:5060;branch=z9hG4bK50603522629637-1157004452658
From: "PCS Phone IL" <sip:+12177665959@209.247.16.5>;tag=VPSF50603522629637
To: <sip:+12179034857@69.1.236.33:5060>
Call-ID: DALMGC0320060930202747006680@209.244.62.68
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@12.206.2.116>
Content-Length: 0
---
-- Executing SIPGetHeader("SIP/2179034857-081b3010", "TO=To") in new stack
-- SIPGetHeader: set variable TO to <sip:+12179034857@69.1.236.33:5060>
-- Executing GotoIf("SIP/2179034857-081b3010", "1?5") in new stack
-- Goto (sipurain,s,5)
-- Executing GotoIf("SIP/2179034857-081b3010", "0?10:6") in new stack
-- Goto (sipurain,s,6)
-- Executing GotoIf("SIP/2179034857-081b3010", "0?10:7") in new stack
-- Goto (sipurain,s,7)
-- Executing Dial("SIP/2179034857-081b3010", "sip/sipura1|30|r") in new stack
We're at 172.21.100.2 port 19026
Adding codec 0x4 (ulaw) to SDP
13 headers, 8 lines
Reliably Transmitting (no NAT) to 172.21.100.10:5060:
INVITE sip:sipura1@172.21.100.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.21.100.2:5060;branch=z9hG4bK6509c353;rport
From: "PCS Phone IL" <sip:+12177665959@172.21.100.2>;tag=as6da68343
To: <sip:sipura1@172.21.100.10:5060>
Contact: <sip:+12177665959@172.21.100.2>
Call-ID: 1ec2c8160c1fa492786bde2b255de2b9@172.21.100.2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 30 Sep 2006 20:27:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 160
v=0
o=root 23965 23965 IN IP4 172.21.100.2
s=session
c=IN IP4 172.21.100.2
t=0 0
m=audio 19026 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
---
-- Called sipura1
Transmitting (no NAT) to 69.1.236.33:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 69.1.236.33;branch=z9hG4bKec6.39b501d4.1;received=69.1.236.33
Via: SIP/2.0/UDP 209.247.16.5:5060;branch=z9hG4bK50603522629637-1157004452658
From: "PCS Phone IL" <sip:+12177665959@209.247.16.5>;tag=VPSF50603522629637
To: <sip:+12179034857@69.1.236.33:5060>;tag=as430782dc
Call-ID: DALMGC0320060930202747006680@209.244.62.68
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@12.206.2.116>
Content-Length: 0
---
<-- SIP read from 172.21.100.10:5060:
SIP/2.0 100 Trying
To: <sip:sipura1@172.21.100.10:5060>
From: "PCS Phone IL" <sip:+12177665959@172.21.100.2>;tag=as6da68343
Call-ID: 1ec2c8160c1fa492786bde2b255de2b9@172.21.100.2
CSeq: 102 INVITE
Via: SIP/2.0/UDP 172.21.100.2:5060;branch=z9hG4bK6509c353
Server: Sipura/SPA2000-2.0.13(g)
Content-Length: 0
--- (8 headers 0 lines)---
<-- SIP read from 172.21.100.10:5060:
SIP/2.0 180 Ringing
To: <sip:sipura1@172.21.100.10:5060>;tag=28532ecf9c6858dci0
From: "PCS Phone IL" <sip:+12177665959@172.21.100.2>;tag=as6da68343
Call-ID: 1ec2c8160c1fa492786bde2b255de2b9@172.21.100.2
CSeq: 102 INVITE
Via: SIP/2.0/UDP 172.21.100.2:5060;branch=z9hG4bK6509c353
Server: Sipura/SPA2000-2.0.13(g)
Content-Length: 0
--- (8 headers 0 lines)---
-- SIP/sipura1-081b8550 is ringing
Destroying call '68ca75e52b6e5efa1364484e5bda5474@172.21.100.2'