<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '045fa3cb6d76323b25cb2d431462e0bd@192.168.32.3' Method: OPTIONS
modi*CLI>
<--- SIP read from 66.54.140.46:5060 --->
INVITE sip:1001@sip.svartalfheim.net SIP/2.0
Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK5db65135;rport
From: "Unknown" <sip:Unknown@66.54.140.46>;tag=as33cf9035
To: <sip:1001@sip.svartalfheim.net>
Contact: <sip:Unknown@66.54.140.46>
Call-ID: 5a56e2a2196f7bdb0cf1130b15af8f0d@66.54.140.46
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 28 Jul 2007 14:51:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
ontent-Type: application/sdp
Content-Length: 306

v=0
o=root 710 710 IN IP4 66.54.140.46
s=session
c=IN IP4 66.54.140.46
t=0 0
m=audio 10418 RTP/AVP 0 8 3 18 101
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:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

<------------->
--- (13 headers 14 lines) ---
Sending to 66.54.140.46 : 5060 (NAT)
Using INVITE request as basis request - 5a56e2a2196f7bdb0cf1130b15af8f0d@66.54.140.46
Found peer '1001'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 66.54.140.46:10418
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format GSM for ID 3
Found description format G729 for ID 18
Found description format telephone-event for ID 101
Capabilities: us - 0x502 (gsm|g729|ilbc), peer - audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x102 (gsm|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 66.54.140.46:10418
Looking for 1001 in incoming (domain sip.svartalfheim.net)
list_route: hop: <sip:Unknown@66.54.140.46>

<--- Transmitting (no NAT) to 66.54.140.46:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK5db65135;received=66.54.140.46;rport=5060
From: "Unknown" <sip:Unknown@66.54.140.46>;tag=as33cf9035
To: <sip:1001@sip.svartalfheim.net>
Call-ID: 5a56e2a2196f7bdb0cf1130b15af8f0d@66.54.140.46
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1001@24.109.78.19>
Content-Length: 0


<------------>
Video is at 192.168.32.3 port 10180
Audio is at 192.168.32.3 port 10036
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x40000 (h261) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.32.20:5060:
INVITE sip:2000@192.168.32.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.32.3:5060;branch=z9hG4bK654ceadf;rport
From: "Unknown" <sip:Unknown@192.168.32.3>;tag=as52c8bfc2
To: <sip:2000@192.168.32.20>
Contact: <sip:Unknown@192.168.32.3>
Call-ID: 369f57517d5e0a9e24668b580e3c9456@192.168.32.3
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 28 Jul 2007 14:51:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 402

v=0
o=root 7904 7904 IN IP4 192.168.32.3
s=session
c=IN IP4 192.168.32.3
b=CT:384
t=0 0
m=audio 10036 RTP/AVP 3 0 8 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 10180 RTP/AVP 31
a=rtpmap:31 H261/90000
a=sendrecv

---

<--- Transmitting (no NAT) to 66.54.140.46:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK5db65135;received=66.54.140.46;rport=5060
From: "Unknown" <sip:Unknown@66.54.140.46>;tag=as33cf9035
To: <sip:1001@sip.svartalfheim.net>;tag=as699eada3
Call-ID: 5a56e2a2196f7bdb0cf1130b15af8f0d@66.54.140.46
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1001@24.109.78.19>
Content-Length: 0


<------------>
modi*CLI>
<--- SIP read from 192.168.32.20:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.32.3:5060;branch=z9hG4bK654ceadf;rport
From: "Unknown" <sip:Unknown@192.168.32.3>;tag=as52c8bfc2
To: <sip:2000@192.168.32.20>;tag=B8F2744C-781701F
CSeq: 102 INVITE
Call-ID: 369f57517d5e0a9e24668b580e3c9456@192.168.32.3
Contact: <sip:2000@192.168.32.20>
User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
modi*CLI>
<--- SIP read from 192.168.32.20:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.32.3:5060;branch=z9hG4bK654ceadf;rport
From: "Unknown" <sip:Unknown@192.168.32.3>;tag=as52c8bfc2
To: <sip:2000@192.168.32.20>;tag=B8F2744C-781701F
CSeq: 102 INVITE
Call-ID: 369f57517d5e0a9e24668b580e3c9456@192.168.32.3
Contact: <sip:2000@192.168.32.20>
User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078
Allow-Events: talk,hold,conference
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
modi*CLI>
<--- SIP read from 192.168.32.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.32.3:5060;branch=z9hG4bK654ceadf;rport
From: "Unknown" <sip:Unknown@192.168.32.3>;tag=as52c8bfc2
To: <sip:2000@192.168.32.20>;tag=B8F2744C-781701F
CSeq: 102 INVITE
Call-ID: 369f57517d5e0a9e24668b580e3c9456@192.168.32.3
Contact: <sip:2000@192.168.32.20>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078
Content-Type: application/sdp
Content-Length: 259

v=0
o=- 1185634268 1185634268 IN IP4 192.168.32.20
s=Polycom IP Phone
c=IN IP4 192.168.32.20
t=0 0
m=audio 2262 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
m=video 0 RTP/AVP 31
a=inactive
a=rtpmap:31 H261/90000

<------------->
--- (11 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found RTP video format 31
Peer audio RTP is at port 192.168.32.20:2262
Found description format PCMU for ID 0
Found description format telephone-event for ID 101
Found description format H261 for ID 31
Capabilities: us - 0x4040e (gsm|ulaw|alaw|ilbc|h261), peer - audio=0x40004 (ulaw|h261)/video=0x40000 (h261), combined - 0x40004 (ulaw|h261)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.32.20:2262
list_route: hop: <sip:2000@192.168.32.20>
set_destination: Parsing <sip:2000@192.168.32.20> for address/port to send to
set_destination: set destination to 192.168.32.20, port 5060
Transmitting (NAT) to 192.168.32.20:5060:
ACK sip:2000@192.168.32.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.32.3:5060;branch=z9hG4bK6b64ef38;rport
From: "Unknown" <sip:Unknown@192.168.32.3>;tag=as52c8bfc2
To: <sip:2000@192.168.32.20>;tag=B8F2744C-781701F
Contact: <sip:Unknown@192.168.32.3>
Call-ID: 369f57517d5e0a9e24668b580e3c9456@192.168.32.3
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Audio is at 24.109.78.19 port 10190
Adding codec 0x2 (gsm) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 66.54.140.46:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK5db65135;received=66.54.140.46;rport=5060
From: "Unknown" <sip:Unknown@66.54.140.46>;tag=as33cf9035
To: <sip:1001@sip.svartalfheim.net>;tag=as699eada3
Call-ID: 5a56e2a2196f7bdb0cf1130b15af8f0d@66.54.140.46
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1001@24.109.78.19>
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 7904 7904 IN IP4 24.109.78.19
s=session
c=IN IP4 24.109.78.19
t=0 0
m=audio 10190 RTP/AVP 3 18 101
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
modi*CLI>
<--- SIP read from 66.54.140.46:5060 --->
ACK sip:1001@24.109.78.19 SIP/2.0
Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK62c38ca9;rport
From: "Unknown" <sip:Unknown@66.54.140.46>;tag=as33cf9035
To: <sip:1001@sip.svartalfheim.net>;tag=as699eada3
Contact: <sip:Unknown@66.54.140.46>
Call-ID: 5a56e2a2196f7bdb0cf1130b15af8f0d@66.54.140.46
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
modi*CLI>
<--- SIP read from 192.168.32.20:5060 --->
BYE sip:Unknown@192.168.32.3 SIP/2.0
Via: SIP/2.0/UDP 192.168.32.20;branch=z9hG4bKd52db911956229AE
From: <sip:2000@192.168.32.20>;tag=B8F2744C-781701F
To: "Unknown" <sip:Unknown@192.168.32.3>;tag=as52c8bfc2
CSeq: 1 BYE
Call-ID: 369f57517d5e0a9e24668b580e3c9456@192.168.32.3
Contact: <sip:2000@192.168.32.20>
User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078
Max-Forwards: 70
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.32.20 : 5060 (NAT)

<--- Transmitting (NAT) to 192.168.32.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.32.20;branch=z9hG4bKd52db911956229AE;received=192.168.32.20
From: <sip:2000@192.168.32.20>;tag=B8F2744C-781701F
To: "Unknown" <sip:Unknown@192.168.32.3>;tag=as52c8bfc2
Call-ID: 369f57517d5e0a9e24668b580e3c9456@192.168.32.3
CSeq: 1 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:Unknown@192.168.32.3>
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '5a56e2a2196f7bdb0cf1130b15af8f0d@66.54.140.46' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:Unknown@66.54.140.46> for address/port to send to
set_destination: set destination to 66.54.140.46, port 5060
Reliably Transmitting (no NAT) to 66.54.140.46:5060:
BYE sip:Unknown@66.54.140.46 SIP/2.0
Via: SIP/2.0/UDP 24.109.78.19:5060;branch=z9hG4bK7c7a46d5;rport
From: <sip:1001@sip.svartalfheim.net>;tag=as699eada3
To: "Unknown" <sip:Unknown@66.54.140.46>;tag=as33cf9035
Call-ID: 5a56e2a2196f7bdb0cf1130b15af8f0d@66.54.140.46
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
modi*CLI>
<--- SIP read from 66.54.140.46:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 24.109.78.19:5060;branch=z9hG4bK7c7a46d5;received=24.109.78.19;rport=5060
From: <sip:1001@sip.svartalfheim.net>;tag=as699eada3
To: "Unknown" <sip:Unknown@66.54.140.46>;tag=as33cf9035
Call-ID: 5a56e2a2196f7bdb0cf1130b15af8f0d@66.54.140.46
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:Unknown@66.54.140.46>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


<------------->
--- (11 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '369f57517d5e0a9e24668b580e3c9456@192.168.32.3' Method: BYE
Really destroying SIP dialog '5a56e2a2196f7bdb0cf1130b15af8f0d@66.54.140.46' Method: ACK
modi*CLI> sip set debug off
SIP Debugging Disabled
modi*CLI>