digit*CLI> sip debug petron
Usage: sip debug
       Enables dumping of SIP packets for debugging purposes

       sip debug ip <host[:PORT]>
       Enables dumping of SIP packets to and from host.

       sip debug peer <peername>
       Enables dumping of SIP packets to and from host.
       Require peer to be registered.
digit*CLI> sip debug peerpetron
Usage: sip debug
       Enables dumping of SIP packets for debugging purposes

       sip debug ip <host[:PORT]>
       Enables dumping of SIP packets to and from host.

       sip debug peer <peername>
       Enables dumping of SIP packets to and from host.
       Require peer to be registered.
digit*CLI> sip debug peer petron
SIP Debugging Enabled for IP: 206.80.91.227:5060
We're at 209.40.224.166 port 61392
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
14 headers, 12 lines
Reliably Transmitting (NAT) to 206.80.91.227:5060:
INVITE sip:19723355333@206.80.91.227 SIP/2.0
Via: SIP/2.0/UDP 209.40.224.166:5060;branch=z9hG4bK1ea159a4;rport
From: "+13143220695" <sip:+13143220695@209.40.224.166>;tag=as0703ebdf
To: <sip:19723355333@206.80.91.227>
Contact: <sip:+13143220695@209.40.224.166>
Call-ID: 3379d6b8278d302f5134d6b77518aec9@209.40.224.166
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "+13143220695" <sip:+13143220695@209.40.224.166>;privacy=off;screen=no
Date: Wed, 13 Jun 2007 22:03:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 16843 16843 IN IP4 209.40.224.166
s=session
c=IN IP4 209.40.224.166
t=0 0
m=audio 61392 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/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 - - - -

---
digit*CLI> 
<-- SIP read from 206.80.91.227:5060: 
SIP/2.0 100 Trying
Call-ID: 3379d6b8278d302f5134d6b77518aec9@209.40.224.166
CSeq: 102 INVITE
From: "+13143220695" <sip:+13143220695@209.40.224.166>;tag=as0703ebdf
To: <sip:19723355333@206.80.91.227>;tag=192.168.1.4+1+2dc0001+502c7a77
Via: SIP/2.0/UDP 209.40.224.166:5060;branch=z9hG4bK1ea159a4
Supported: replaces
Supported: 100rel
Contact: <sip:19723355333@206.80.91.227:5060>
Content-Length: 0
User-Agent: TeleVantage-UA/7.50.4880
Server: TeleVantage-UA/7.50.4880


--- (12 headers 0 lines) ---
digit*CLI> 
<-- SIP read from 206.80.91.227:5060: 
SIP/2.0 200 OK
Call-ID: 3379d6b8278d302f5134d6b77518aec9@209.40.224.166
CSeq: 102 INVITE
From: "+13143220695" <sip:+13143220695@209.40.224.166>;tag=as0703ebdf
To: <sip:19723355333@206.80.91.227>;tag=192.168.1.4+1+2dc0001+502c7a77
Via: SIP/2.0/UDP 209.40.224.166:5060;branch=z9hG4bK1ea159a4
Allow-Events: message-summary
Allow-Events: presence
Allow-Events: refer
Supported: replaces
Supported: 100rel
Allow: ACK
Allow: BYE
Allow: CANCEL
Allow: INFO
Allow: INVITE
Allow: NOTIFY
Allow: OPTIONS
Allow: PRACK
Allow: REFER
Allow: SUBSCRIBE
Allow: UPDATE
Contact: <sip:19723355333@206.80.91.227:5060>
Content-Type: application/sdp
Content-Length: 193
User-Agent: TeleVantage-UA/7.50.4880
Server: TeleVantage-UA/7.50.4880

v=0
o=TeleVantage 554 0 IN IP4 206.80.91.227
s=phone-call
c=IN IP4 206.80.91.227
t=0 0
m=audio 6004 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000/1
a=rtpmap:101 telephone-event/8000/1
a=ptime:20

--- (27 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 206.80.91.227:6004
Found description format pcmu
Found description format telephone-event
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:19723355333@206.80.91.227:5060>
set_destination: Parsing <sip:19723355333@206.80.91.227:5060> for address/port to send to
set_destination: set destination to 206.80.91.227, port 5060
Transmitting (NAT) to 206.80.91.227:5060:
ACK sip:19723355333@206.80.91.227:5060 SIP/2.0
Via: SIP/2.0/UDP 209.40.224.166:5060;branch=z9hG4bK2f0505c6;rport
From: "+13143220695" <sip:+13143220695@209.40.224.166>;tag=as0703ebdf
To: <sip:19723355333@206.80.91.227>;tag=192.168.1.4+1+2dc0001+502c7a77
Contact: <sip:+13143220695@209.40.224.166>
Call-ID: 3379d6b8278d302f5134d6b77518aec9@209.40.224.166
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "+13143220695" <sip:+13143220695@209.40.224.166>;privacy=off;screen=no
Content-Length: 0


---
set_destination: Parsing <sip:19723355333@206.80.91.227:5060> for address/port to send to
set_destination: set destination to 206.80.91.227, port 5060
We're at 209.40.224.166 port 61392
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
14 headers, 12 lines
Reliably Transmitting (NAT) to 206.80.91.227:5060:
INVITE sip:19723355333@206.80.91.227:5060 SIP/2.0
Via: SIP/2.0/UDP 209.40.224.166:5060;branch=z9hG4bK2e9979c4;rport
From: "+13143220695" <sip:+13143220695@209.40.224.166>;tag=as0703ebdf
To: <sip:19723355333@206.80.91.227>;tag=192.168.1.4+1+2dc0001+502c7a77
Contact: <sip:+13143220695@209.40.224.166>
Call-ID: 3379d6b8278d302f5134d6b77518aec9@209.40.224.166
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "+13143220695" <sip:+13143220695@209.40.224.166>;privacy=off;screen=no
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 16843 16844 IN IP4 63.215.29.83
s=session
c=IN IP4 63.215.29.83
t=0 0
m=audio 60010 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/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 - - - -

---
digit*CLI> 
<-- SIP read from 206.80.91.227:5060: 
SIP/2.0 100 Trying
Call-ID: 3379d6b8278d302f5134d6b77518aec9@209.40.224.166
CSeq: 103 INVITE
From: "+13143220695" <sip:+13143220695@209.40.224.166>;tag=as0703ebdf
To: <sip:19723355333@206.80.91.227>;tag=192.168.1.4+1+2dc0001+502c7a77
Via: SIP/2.0/UDP 209.40.224.166:5060;branch=z9hG4bK2e9979c4
Supported: replaces
Supported: 100rel
Contact: <sip:19723355333@206.80.91.227:5060>
Content-Length: 0
User-Agent: TeleVantage-UA/7.50.4880
Server: TeleVantage-UA/7.50.4880


--- (12 headers 0 lines) ---
digit*CLI> 
<-- SIP read from 206.80.91.227:5060: 
SIP/2.0 200 OK
Call-ID: 3379d6b8278d302f5134d6b77518aec9@209.40.224.166
CSeq: 103 INVITE
From: "+13143220695" <sip:+13143220695@209.40.224.166>;tag=as0703ebdf
To: <sip:19723355333@206.80.91.227>;tag=192.168.1.4+1+2dc0001+502c7a77
Via: SIP/2.0/UDP 209.40.224.166:5060;branch=z9hG4bK2e9979c4
Allow-Events: message-summary
Allow-Events: presence
Allow-Events: refer
Supported: replaces
Supported: 100rel
Allow: ACK
Allow: BYE
Allow: CANCEL
Allow: INFO
Allow: INVITE
Allow: NOTIFY
Allow: OPTIONS
Allow: PRACK
Allow: REFER
Allow: SUBSCRIBE
Allow: UPDATE
Contact: <sip:19723355333@206.80.91.227:5060>
Content-Type: application/sdp
Content-Length: 193
User-Agent: TeleVantage-UA/7.50.4880
Server: TeleVantage-UA/7.50.4880

v=0
o=TeleVantage 555 0 IN IP4 206.80.91.227
s=phone-call
c=IN IP4 206.80.91.227
t=0 0
m=audio 6004 RTP/AVP 0 101
a=rtpmap:0 pcmu/8000/1
a=rtpmap:101 telephone-event/8000/1
a=ptime:20

--- (27 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 206.80.91.227:6004
Found description format pcmu
Found description format telephone-event
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:19723355333@206.80.91.227:5060> for address/port to send to
set_destination: set destination to 206.80.91.227, port 5060
Transmitting (NAT) to 206.80.91.227:5060:
ACK sip:19723355333@206.80.91.227:5060 SIP/2.0
Via: SIP/2.0/UDP 209.40.224.166:5060;branch=z9hG4bK59dda71a;rport
From: "+13143220695" <sip:+13143220695@209.40.224.166>;tag=as0703ebdf
To: <sip:19723355333@206.80.91.227>;tag=192.168.1.4+1+2dc0001+502c7a77
Contact: <sip:+13143220695@209.40.224.166>
Call-ID: 3379d6b8278d302f5134d6b77518aec9@209.40.224.166
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "+13143220695" <sip:+13143220695@209.40.224.166>;privacy=off;screen=no
Content-Length: 0


---
set_destination: Parsing <sip:19723355333@206.80.91.227:5060> for address/port to send to
set_destination: set destination to 206.80.91.227, port 5060
We're at 209.40.224.166 port 61392
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
14 headers, 10 lines
Reliably Transmitting (NAT) to 206.80.91.227:5060:
INVITE sip:19723355333@206.80.91.227:5060 SIP/2.0
Via: SIP/2.0/UDP 209.40.224.166:5060;branch=z9hG4bK7e5dfee1;rport
From: "+13143220695" <sip:+13143220695@209.40.224.166>;tag=as0703ebdf
To: <sip:19723355333@206.80.91.227>;tag=192.168.1.4+1+2dc0001+502c7a77
Contact: <sip:+13143220695@209.40.224.166>
Call-ID: 3379d6b8278d302f5134d6b77518aec9@209.40.224.166
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "+13143220695" <sip:+13143220695@209.40.224.166>;privacy=off;screen=no
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-asterisk-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 220

v=0it*CLI> 
o=root 16843 16845 IN IP4 209.40.224.166
s=session
c=IN IP4 209.40.224.166
t=0 0
m=audio 61392 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Scheduling destruction of call '3379d6b8278d302f5134d6b77518aec9@209.40.224.166' in 32000 ms
digit*CLI> 
<-- SIP read from 206.80.91.227:5060: 
SIP/2.0 100 Trying
Call-ID: 3379d6b8278d302f5134d6b77518aec9@209.40.224.166
CSeq: 104 INVITE
From: "+13143220695" <sip:+13143220695@209.40.224.166>;tag=as0703ebdf
To: <sip:19723355333@206.80.91.227>;tag=192.168.1.4+1+2dc0001+502c7a77
Via: SIP/2.0/UDP 209.40.224.166:5060;branch=z9hG4bK7e5dfee1
Supported: replaces
Supported: 100rel
Contact: <sip:19723355333@206.80.91.227:5060>
Content-Length: 0
User-Agent: TeleVantage-UA/7.50.4880
Server: TeleVantage-UA/7.50.4880


--- (12 headers 0 lines) ---
set_destination: Parsing <sip:19723355333@206.80.91.227:5060> for address/port to send to
set_destination: set destination to 206.80.91.227, port 5060
Reliably Transmitting (NAT) to 206.80.91.227:5060:
BYE sip:19723355333@206.80.91.227:5060 SIP/2.0
Via: SIP/2.0/UDP 209.40.224.166:5060;branch=z9hG4bK190f1913;rport
From: "+13143220695" <sip:+13143220695@209.40.224.166>;tag=as0703ebdf
To: <sip:19723355333@206.80.91.227>;tag=192.168.1.4+1+2dc0001+502c7a77
Call-ID: 3379d6b8278d302f5134d6b77518aec9@209.40.224.166
CSeq: 105 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "+13143220695" <sip:+13143220695@209.40.224.166>;privacy=off;screen=no
Content-Length: 0


---
Scheduling destruction of call '3379d6b8278d302f5134d6b77518aec9@209.40.224.166' in 32000 ms
digit*CLI> 
<-- SIP read from 206.80.91.227:5060: 
SIP/2.0 200 OK
Call-ID: 3379d6b8278d302f5134d6b77518aec9@209.40.224.166
CSeq: 105 BYE
From: "+13143220695" <sip:+13143220695@209.40.224.166>;tag=as0703ebdf
To: <sip:19723355333@206.80.91.227>;tag=192.168.1.4+1+2dc0001+502c7a77
Via: SIP/2.0/UDP 209.40.224.166:5060;branch=z9hG4bK190f1913
Allow-Events: message-summary
Allow-Events: presence
Allow-Events: refer
Supported: replaces
Supported: 100rel
Content-Length: 0
User-Agent: TeleVantage-UA/7.50.4880
Server: TeleVantage-UA/7.50.4880


--- (14 headers 0 lines) ---
digit*CLI> 
<-- SIP read from 206.80.91.227:5060: 
OPTIONS sip:209.40.224.166 SIP/2.0
Via: SIP/2.0/UDP 206.80.91.227:5060;branch=z9hG4bK+52fa438a0c94f0c0c06af02083009e8e+192.168.1.4+1
Allow-Events: message-summary
Allow-Events: presence
Allow-Events: refer
Max-Forwards: 70
Call-ID: 7F0AB8DE@206.80.91.227
From: <sip:192.168.1.4>;tag=192.168.1.4+1+0+9aa291e8
CSeq: 270095357 OPTIONS
Supported: replaces
Supported: 100rel
Content-Length: 0
To: sip:209.40.224.166
Contact: <sip:206.80.91.227:5060>
User-Agent: TeleVantage-UA/7.50.4880


--- (15 headers 0 lines) ---
Looking for s in default (domain 209.40.224.166)
Transmitting (no NAT) to 206.80.91.227:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 206.80.91.227:5060;branch=z9hG4bK+52fa438a0c94f0c0c06af02083009e8e+192.168.1.4+1;received=206.80.91.227
From: <sip:192.168.1.4>;tag=192.168.1.4+1+0+9aa291e8
To: sip:209.40.224.166;tag=as09a53939
Call-ID: 7F0AB8DE@206.80.91.227
CSeq: 270095357 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp
Content-Length: 0


---
Destroying call '7F0AB8DE@206.80.91.227'