rendered paste body<-- SIP read from 67.55.159.160:5060:
INVITE sip:6418272550@192.168.1.3:5060 SIP/2.0
Record-Route: <sip:67.55.159.160;ftag=as03209cec;lr=on>
Via: SIP/2.0/UDP 67.55.159.160;branch=z9hG4bK543a.1b250865.0
Via: SIP/2.0/UDP 67.55.159.156:5060;branch=z9hG4bK21a7138d;rport=5060
From: "8183008950" <sip:8183008950@67.55.159.156>;tag=as03209cec
To: <sip:6418272550@sipnumber.net>
Contact: <sip:8183008950@67.55.159.156:5060>
Call-ID: 5cb1dee07292e55442709e1d6e0bd8c9@67.55.159.156
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 16
Date: Thu, 21 Dec 2006 21:47:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 441
v=0
o=root 16494 16494 IN IP4 67.55.159.156
s=session
c=IN IP4 67.55.159.156
t=0 0
m=audio 19612 RTP/AVP 0 8 3 18 111 5 10 7 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (15 headers 19 lines)---
Using INVITE request as basis request - 5cb1dee07292e55442709e1d6e0bd8c9@67.55.159.156
Sending to 67.55.159.160 : 5060 (NAT)
Found peer 'freedigits_Out'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 111
Found RTP audio format 5
Found RTP audio format 10
Found RTP audio format 7
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 67.55.159.156:19612
Found description format PCMU
Found description format PCMA
Found description format GSM
Found description format G729
Found description format G726-32
Found description format DVI4
Found description format L16
Found description format LPC
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x5fe (gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 6418272550 in from-pstn (domain 192.168.1.3)
list_route: hop: <sip:67.55.159.160;ftag=as03209cec;lr=on>
Transmitting (NAT) to 67.55.159.160:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 67.55.159.160;branch=z9hG4bK543a.1b250865.0;received=67.55.159.160
Via: SIP/2.0/UDP 67.55.159.156:5060;branch=z9hG4bK21a7138d;rport=5060
From: "8183008950" <sip:8183008950@67.55.159.156>;tag=as03209cec
To: <sip:6418272550@sipnumber.net>
Call-ID: 5cb1dee07292e55442709e1d6e0bd8c9@67.55.159.156
CSeq: 102 INVITE
User-Agent: Linksys/SPA2102-3.3.5(a)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6418272550@192.168.1.3>
Content-Length: 0
---
-- Executing Set("SIP/6418272550-09c27710", "FROM_DID=6418272550") in new stack
-- Executing Gosub("SIP/6418272550-09c27710", "app-blacklist-check|s|1") in new stack
-- Executing LookupBlacklist("SIP/6418272550-09c27710", "") in new stack
-- Executing GotoIf("SIP/6418272550-09c27710", "0?blacklisted") in new stack
-- Executing Return("SIP/6418272550-09c27710", "") in new stack
-- Executing Goto("SIP/6418272550-09c27710", "app-blackhole|busy|1") in new stack
-- Goto (app-blackhole,busy,1)
-- Executing NoOp("SIP/6418272550-09c27710", "Blackhole Dest: Busy") in new stack
-- Executing Answer("SIP/6418272550-09c27710", "") in new stack
We're at 192.168.1.3 port 14126
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 67.55.159.160:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.55.159.160;branch=z9hG4bK543a.1b250865.0;received=67.55.159.160
Via: SIP/2.0/UDP 67.55.159.156:5060;branch=z9hG4bK21a7138d;rport=5060
Record-Route: <sip:67.55.159.160;ftag=as03209cec;lr=on>
From: "8183008950" <sip:8183008950@67.55.159.156>;tag=as03209cec
To: <sip:6418272550@sipnumber.net>;tag=as1e247133
Call-ID: 5cb1dee07292e55442709e1d6e0bd8c9@67.55.159.156
CSeq: 102 INVITE
User-Agent: Linksys/SPA2102-3.3.5(a)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6418272550@192.168.1.3>
Content-Type: application/sdp
Content-Length: 259
v=0
o=root 2741 2741 IN IP4 192.168.1.3
s=session
c=IN IP4 192.168.1.3
t=0 0
m=audio 14126 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Executing PlayTones("SIP/6418272550-09c27710", "busy") in new stack
-- Executing Busy("SIP/6418272550-09c27710", "") in new stack
asterisk1*CLI>
<-- SIP read from 67.55.159.160:5060:
ACK sip:6418272550@192.168.1.3:5060 SIP/2.0
Via: SIP/2.0/UDP 67.55.159.160;branch=0
Via: SIP/2.0/UDP 67.55.159.156:5060;branch=z9hG4bK50c065ab;rport=5060
From: "8183008950" <sip:8183008950@67.55.159.156>;tag=as03209cec
To: <sip:6418272550@sipnumber.net>;tag=as1e247133
Contact: <sip:8183008950@67.55.159.156:5060>
Call-ID: 5cb1dee07292e55442709e1d6e0bd8c9@67.55.159.156
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 16
Content-Length: 0
--- (11 headers 0 lines)---