rendered paste bodypbx*CLI> sip show peer jon
pbx*CLI>
* Name : jon
Secret : <Set>
MD5Secret : <Not set>
Context : local
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox : 7102,3149
VM Extension : asterisk
LastMsgsSent : 0
Call limit : 0
Dynamic : Yes
Callerid : "Jon Webster" <3149>
Expire : 3455
Insecure : no
Nat : Always
ACL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
Trust RPID : No
Send RPID : No
DTMFmode : rfc2833
LastMsg : 0
ToHost :
Addr->IP : 69.69.144.77 Port 5060
Defaddr->IP : 0.0.0.0 Port 5060
Def. Username: jon
SIP Options : (none)
Codecs : 0x80006 (gsm|ulaw|h263)
Codec Order : (ulaw,gsm,h263)
Status : OK (103 ms)
Useragent : Sipura/SPA941-4.1.8
Reg. Contact : sip:jon@192.168.22.16:5060
pbx*CLI> sip debug peer jon
SIP Debugging Enabled for IP: 69.69.144.77:5060
pbx*CLI> dial 3149
pbx*CLI>
<-- SIP read from 69.69.144.77:5060:
NOTIFY sip:pbx.elephantoutlook.com SIP/2.0
Via: SIP/2.0/UDP 192.168.22.16:5060;branch=z9hG4bK-9b3d4203
From: "Jon Webster <3149>" <sip:jon@pbx.elephantoutlook.com>;tag=5de79c897e369923o0
To: <sip:pbx.elephantoutlook.com>
Call-ID: 37a70dc9-807bd8e3@192.168.22.16
CSeq: 19 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA941-4.1.8
Content-Length: 0
--- (10 headers 0 lines)---
Transmitting (no NAT) to 69.69.144.77:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.22.16:5060;branch=z9hG4bK-9b3d4203;received=69.69.144.77
From: "Jon Webster <3149>" <sip:jon@pbx.elephantoutlook.com>;tag=5de79c897e369923o0
To: <sip:pbx.elephantoutlook.com>;tag=as0d5a0d1f
Call-ID: 37a70dc9-807bd8e3@192.168.22.16
CSeq: 19 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Destroying call '37a70dc9-807bd8e3@192.168.22.16'
pbx*CLI> hangup
set_destination: Parsing <3149> for address/port to send to
Sep 14 22:04:01 WARNING[17864]: chan_sip.c:4029 set_destination: Can't find address for host '3149'
Reliably Transmitting (NAT) to 69.69.144.77:5060:
BYE 3149 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK6a743bd9;rport
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as357a17e8
To: <sip:jon@192.168.22.16:5060>;tag=d778c4b3d3be8f86i0
Contact: <sip:asterisk@66.192.107.225>
Call-ID: 165b0baa16841da57b4e9bf4176d50bc@66.192.107.225
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
<< Hangup on console >>
Retransmitting #1 (NAT) to 69.69.144.77:5060:
BYE 3149 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK6a743bd9;rport
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as357a17e8
To: <sip:jon@192.168.22.16:5060>;tag=d778c4b3d3be8f86i0
Contact: <sip:asterisk@66.192.107.225>
Call-ID: 165b0baa16841da57b4e9bf4176d50bc@66.192.107.225
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Retransmitting #2 (NAT) to 69.69.144.77:5060:
BYE 3149 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK6a743bd9;rport
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as357a17e8
To: <sip:jon@192.168.22.16:5060>;tag=d778c4b3d3be8f86i0
Contact: <sip:asterisk@66.192.107.225>
Call-ID: 165b0baa16841da57b4e9bf4176d50bc@66.192.107.225
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Retransmitting #3 (NAT) to 69.69.144.77:5060:
BYE 3149 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK6a743bd9;rport
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as357a17e8
To: <sip:jon@192.168.22.16:5060>;tag=d778c4b3d3be8f86i0
Contact: <sip:asterisk@66.192.107.225>
Call-ID: 165b0baa16841da57b4e9bf4176d50bc@66.192.107.225
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
12 headers, 0 lines
Reliably Transmitting (NAT) to 69.69.144.77:5060:
OPTIONS sip:jon@192.168.22.16:5060 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK7c43d155;rport
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as71017e3c
To: <sip:jon@192.168.22.16:5060>
Contact: <sip:asterisk@66.192.107.225>
Call-ID: 4185c8eb7a5a03ed7041b1d534d88c6e@66.192.107.225
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 15 Sep 2006 02:04:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
pbx*CLI>
<-- SIP read from 69.69.144.77:5060:
SIP/2.0 200 OK
To: <sip:jon@192.168.22.16:5060>;tag=cc407501a462db0bi0
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as71017e3c
Call-ID: 4185c8eb7a5a03ed7041b1d534d88c6e@66.192.107.225
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK7c43d155
Server: Sipura/SPA941-4.1.8
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
--- (9 headers 0 lines)---
Destroying call '4185c8eb7a5a03ed7041b1d534d88c6e@66.192.107.225'
Retransmitting #4 (NAT) to 69.69.144.77:5060:
BYE 3149 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK6a743bd9;rport
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as357a17e8
To: <sip:jon@192.168.22.16:5060>;tag=d778c4b3d3be8f86i0
Contact: <sip:asterisk@66.192.107.225>
Call-ID: 165b0baa16841da57b4e9bf4176d50bc@66.192.107.225
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
pbx*CLI> hangup
No call to hang up
pbx*CLI>
<-- SIP read from 69.69.144.77:5060:
NOTIFY sip:pbx.elephantoutlook.com SIP/2.0
Via: SIP/2.0/UDP 192.168.22.16:5060;branch=z9hG4bK-319c17f2
From: "Jon Webster <3149>" <sip:jon@pbx.elephantoutlook.com>;tag=5de79c897e369923o0
To: <sip:pbx.elephantoutlook.com>
Call-ID: 37a70dc9-807bd8e3@192.168.22.16
CSeq: 20 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA941-4.1.8
Content-Length: 0
--- (10 headers 0 lines)---
Transmitting (no NAT) to 69.69.144.77:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.22.16:5060;branch=z9hG4bK-319c17f2;received=69.69.144.77
From: "Jon Webster <3149>" <sip:jon@pbx.elephantoutlook.com>;tag=5de79c897e369923o0
To: <sip:pbx.elephantoutlook.com>;tag=as34a54103
Call-ID: 37a70dc9-807bd8e3@192.168.22.16
CSeq: 20 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Destroying call '37a70dc9-807bd8e3@192.168.22.16'
Retransmitting #5 (NAT) to 69.69.144.77:5060:
BYE 3149 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK6a743bd9;rport
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as357a17e8
To: <sip:jon@192.168.22.16:5060>;tag=d778c4b3d3be8f86i0
Contact: <sip:asterisk@66.192.107.225>
Call-ID: 165b0baa16841da57b4e9bf4176d50bc@66.192.107.225
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Retransmitting #6 (NAT) to 69.69.144.77:5060:
BYE 3149 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK6a743bd9;rport
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as357a17e8
To: <sip:jon@192.168.22.16:5060>;tag=d778c4b3d3be8f86i0
Contact: <sip:asterisk@66.192.107.225>
Call-ID: 165b0baa16841da57b4e9bf4176d50bc@66.192.107.225
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Sep 14 22:04:12 WARNING[3730]: chan_sip.c:1226 retrans_pkt: Maximum retries exceeded on transmission 165b0baa16841da57b4e9bf4176d50bc@66.192.107.225 for seqno 103 (Non-critical Request)
Destroying call '165b0baa16841da57b4e9bf4176d50bc@66.192.107.225'
pbx*CLI>
<-- SIP read from 69.69.144.77:5060:
NOTIFY sip:pbx.elephantoutlook.com SIP/2.0
Via: SIP/2.0/UDP 192.168.22.16:5060;branch=z9hG4bK-117628b5
From: "Jon Webster <3149>" <sip:jon@pbx.elephantoutlook.com>;tag=5de79c897e369923o0
To: <sip:pbx.elephantoutlook.com>
Call-ID: 37a70dc9-807bd8e3@192.168.22.16
CSeq: 21 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA941-4.1.8
Content-Length: 0
--- (10 headers 0 lines)---
Transmitting (no NAT) to 69.69.144.77:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.22.16:5060;branch=z9hG4bK-117628b5;received=69.69.144.77
From: "Jon Webster <3149>" <sip:jon@pbx.elephantoutlook.com>;tag=5de79c897e369923o0
To: <sip:pbx.elephantoutlook.com>;tag=as0991c29d
Call-ID: 37a70dc9-807bd8e3@192.168.22.16
CSeq: 21 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Destroying call '37a70dc9-807bd8e3@192.168.22.16'
pbx*CLI> dial 3149
-- Executing Goto("OSS/dsp", "jon|1") in new stack
-- Goto (local,jon,1)
-- Executing Macro("OSS/dsp", "stdexten|3149|SIP/jon&SIP/jonlaptop") in new stack
-- Executing Dial("OSS/dsp", "SIP/jon&SIP/jonlaptop|20|tw") in new stack
We're at 66.192.107.225 port 15676
Video is at 66.192.107.225 port 11668
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x80000 (h263) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 13 lines
Reliably Transmitting (NAT) to 69.69.144.77:5060:
INVITE sip:jon@192.168.22.16:5060 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK7875712a;rport
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as342ff86c
To: <sip:jon@192.168.22.16:5060>
Contact: <sip:asterisk@66.192.107.225>
Call-ID: 60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 15 Sep 2006 02:04:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 291
v=0
o=root 3716 3716 IN IP4 66.192.107.225
s=session
c=IN IP4 66.192.107.225
t=0 0
m=audio 15676 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
m=video 11668 RTP/AVP 34
a=rtpmap:34 H263/90000
---
-- Called jon
Sep 14 22:04:27 NOTICE[17877]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
pbx*CLI>
<-- SIP read from 69.69.144.77:5060:
SIP/2.0 100 Trying
To: <sip:jon@192.168.22.16:5060>
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as342ff86c
Call-ID: 60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225
CSeq: 102 INVITE
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK7875712a
Server: Sipura/SPA941-4.1.8
Content-Length: 0
--- (8 headers 0 lines)---
pbx*CLI>
<-- SIP read from 69.69.144.77:5060:
SIP/2.0 180 Ringing
To: <sip:jon@192.168.22.16:5060>;tag=289819e859262251i0
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as342ff86c
Call-ID: 60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225
CSeq: 102 INVITE
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK7875712a
Server: Sipura/SPA941-4.1.8
Content-Length: 0
--- (8 headers 0 lines)---
-- SIP/jon-081ebfd8 is ringing
Sep 14 22:04:27 WARNING[3723]: chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
pbx*CLI>
<-- SIP read from 69.69.144.77:5060:
SIP/2.0 200 OK
To: <sip:jon@192.168.22.16:5060>;tag=289819e859262251i0
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as342ff86c
Call-ID: 60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225
CSeq: 102 INVITE
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK7875712a
Contact: "Jon Webster <3149>" <sip:jon@192.168.22.16:5060>
Server: Sipura/SPA941-4.1.8
Content-Length: 206
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp
v=0
o=- 32922 32922 IN IP4 192.168.22.16
s=-
c=IN IP4 192.168.22.16
t=0 0
m=audio 16392 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (11 headers 11 lines)---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.22.16:16392
Peer video RTP is at port 192.168.22.16:65535
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x80006 (gsm|ulaw|h263), 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: <3149>
set_destination: Parsing <3149> for address/port to send to
Sep 14 22:04:29 WARNING[3730]: chan_sip.c:4029 set_destination: Can't find address for host '3149'
Transmitting (NAT) to 69.69.144.77:5060:
ACK 3149 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK5ca61f19;rport
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as342ff86c
To: <sip:jon@192.168.22.16:5060>;tag=289819e859262251i0
Contact: <sip:asterisk@66.192.107.225>
Call-ID: 60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225
CSeq: 102 ACK
User-Agent: Asterisk PBX
ax-Forwards: 70
Content-Length: 0
---
-- SIP/jon-081ebfd8 answered OSS/dsp
<< Console call has been answered >>
pbx*CLI>
<-- SIP read from 69.69.144.77:5060:
SIP/2.0 200 OK
To: <sip:jon@192.168.22.16:5060>;tag=289819e859262251i0
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as342ff86c
Call-ID: 60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225
CSeq: 102 INVITE
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK7875712a
Contact: "Jon Webster <3149>" <sip:jon@192.168.22.16:5060>
Server: Sipura/SPA941-4.1.8
Content-Length: 206
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp
v=0
o=- 32922 32922 IN IP4 192.168.22.16
s=-
c=IN IP4 192.168.22.16
t=0 0
m=audio 16392 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (11 headers 11 lines)---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.22.16:16392
Peer video RTP is at port 192.168.22.16:65535
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x80006 (gsm|ulaw|h263), 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 <3149> for address/port to send to
Sep 14 22:04:30 WARNING[3730]: chan_sip.c:4029 set_destination: Can't find address for host '3149'
Transmitting (NAT) to 69.69.144.77:5060:
ACK 3149 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK6b8258bc;rport
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as342ff86c
To: <sip:jon@192.168.22.16:5060>;tag=289819e859262251i0
Contact: <sip:asterisk@66.192.107.225>
Call-ID: 60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
pbx*CLI>
<-- SIP read from 69.69.144.77:5060:
SIP/2.0 200 OK
To: <sip:jon@192.168.22.16:5060>;tag=289819e859262251i0
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as342ff86c
Call-ID: 60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225
CSeq: 102 INVITE
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK7875712a
Contact: "Jon Webster <3149>" <sip:jon@192.168.22.16:5060>
Server: Sipura/SPA941-4.1.8
Content-Length: 206
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp
v=0
o=- 32922 32922 IN IP4 192.168.22.16
s=-
c=IN IP4 192.168.22.16
t=0 0
m=audio 16392 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (11 headers 11 lines)---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.22.16:16392
Peer video RTP is at port 192.168.22.16:65535
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x80006 (gsm|ulaw|h263), 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 <3149> for address/port to send to
Sep 14 22:04:31 WARNING[3730]: chan_sip.c:4029 set_destination: Can't find address for host '3149'
Transmitting (NAT) to 69.69.144.77:5060:
ACK 3149 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK4d5e7977;rport
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as342ff86c
To: <sip:jon@192.168.22.16:5060>;tag=289819e859262251i0
Contact: <sip:asterisk@66.192.107.225>
Call-ID: 60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
pbx*CLI>
<-- SIP read from 69.69.144.77:5060:
SIP/2.0 200 OK
To: <sip:jon@192.168.22.16:5060>;tag=289819e859262251i0
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as342ff86c
Call-ID: 60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225
CSeq: 102 INVITE
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK7875712a
Contact: "Jon Webster <3149>" <sip:jon@192.168.22.16:5060>
Server: Sipura/SPA941-4.1.8
Content-Length: 206
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp
v=0
o=- 32922 32922 IN IP4 192.168.22.16
s=-
c=IN IP4 192.168.22.16
t=0 0
m=audio 16392 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (11 headers 11 lines)---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.22.16:16392
Peer video RTP is at port 192.168.22.16:65535
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x80006 (gsm|ulaw|h263), 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 <3149> for address/port to send to
Sep 14 22:04:33 WARNING[3730]: chan_sip.c:4029 set_destination: Can't find address for host '3149'
Transmitting (NAT) to 69.69.144.77:5060:
ACK 3149 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK482d3102;rport
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as342ff86c
To: <sip:jon@192.168.22.16:5060>;tag=289819e859262251i0
Contact: <sip:asterisk@66.192.107.225>
Call-ID: 60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
pbx*CLI>
<-- SIP read from 69.69.144.77:5060:
NOTIFY sip:pbx.elephantoutlook.com SIP/2.0
Via: SIP/2.0/UDP 192.168.22.16:5060;branch=z9hG4bK-d1043275
From: "Jon Webster <3149>" <sip:jon@pbx.elephantoutlook.com>;tag=5de79c897e369923o0
To: <sip:pbx.elephantoutlook.com>
Call-ID: 37a70dc9-807bd8e3@192.168.22.16
CSeq: 22 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA941-4.1.8
Content-Length: 0
--- (10 headers 0 lines)---
Transmitting (no NAT) to 69.69.144.77:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.22.16:5060;branch=z9hG4bK-d1043275;received=69.69.144.77
From: "Jon Webster <3149>" <sip:jon@pbx.elephantoutlook.com>;tag=5de79c897e369923o0
To: <sip:pbx.elephantoutlook.com>;tag=as526ac43d
Call-ID: 37a70dc9-807bd8e3@192.168.22.16
CSeq: 22 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Destroying call '37a70dc9-807bd8e3@192.168.22.16'
pbx*CLI>
<-- SIP read from 69.69.144.77:5060:
SIP/2.0 200 OK
To: <sip:jon@192.168.22.16:5060>;tag=289819e859262251i0
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as342ff86c
Call-ID: 60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225
CSeq: 102 INVITE
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK7875712a
Contact: "Jon Webster <3149>" <sip:jon@192.168.22.16:5060>
Server: Sipura/SPA941-4.1.8
Content-Length: 206
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp
v=0
o=- 32922 32922 IN IP4 192.168.22.16
s=-
c=IN IP4 192.168.22.16
t=0 0
m=audio 16392 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (11 headers 11 lines)---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.22.16:16392
Peer video RTP is at port 192.168.22.16:65535
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x80006 (gsm|ulaw|h263), 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 <3149> for address/port to send to
Sep 14 22:04:37 WARNING[3730]: chan_sip.c:4029 set_destination: Can't find address for host '3149'
Transmitting (NAT) to 69.69.144.77:5060:
ACK 3149 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK55637c6f;rport
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as342ff86c
To: <sip:jon@192.168.22.16:5060>;tag=289819e859262251i0
Contact: <sip:asterisk@66.192.107.225>
Call-ID: 60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
pbx*CLI>
<-- SIP read from 69.69.144.77:5060:
SIP/2.0 200 OK
To: <sip:jon@192.168.22.16:5060>;tag=289819e859262251i0
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as342ff86c
Call-ID: 60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225
CSeq: 102 INVITE
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK7875712a
Contact: "Jon Webster <3149>" <sip:jon@192.168.22.16:5060>
Server: Sipura/SPA941-4.1.8
Content-Length: 206
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp
v=0
o=- 32922 32922 IN IP4 192.168.22.16
s=-
c=IN IP4 192.168.22.16
t=0 0
m=audio 16392 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (11 headers 11 lines)---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.22.16:16392
Peer video RTP is at port 192.168.22.16:65535
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x80006 (gsm|ulaw|h263), 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 <3149> for address/port to send to
Sep 14 22:04:41 WARNING[3730]: chan_sip.c:4029 set_destination: Can't find address for host '3149'
Transmitting (NAT) to 69.69.144.77:5060:
ACK 3149 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK3b16f6fd;rport
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as342ff86c
To: <sip:jon@192.168.22.16:5060>;tag=289819e859262251i0
Contact: <sip:asterisk@66.192.107.225>
Call-ID: 60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
pbx*CLI> hangup
set_destination: Parsing <3149> for address/port to send to
Sep 14 22:04:43 WARNING[17877]: chan_sip.c:4029 set_destination: Can't find address for host '3149'
Reliably Transmitting (NAT) to 69.69.144.77:5060:
BYE 3149 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK30b4d8dc;rport
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as342ff86c
To: <sip:jon@192.168.22.16:5060>;tag=289819e859262251i0
Contact: <sip:asterisk@66.192.107.225>
Call-ID: 60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
<< Hangup on console >>
Retransmitting #1 (NAT) to 69.69.144.77:5060:
BYE 3149 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK30b4d8dc;rport
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as342ff86c
To: <sip:jon@192.168.22.16:5060>;tag=289819e859262251i0
Contact: <sip:asterisk@66.192.107.225>
Call-ID: 60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Retransmitting #2 (NAT) to 69.69.144.77:5060:
BYE 3149 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK30b4d8dc;rport
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as342ff86c
To: <sip:jon@192.168.22.16:5060>;tag=289819e859262251i0
Contact: <sip:asterisk@66.192.107.225>
Call-ID: 60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Retransmitting #3 (NAT) to 69.69.144.77:5060:
BYE 3149 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK30b4d8dc;rport
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as342ff86c
To: <sip:jon@192.168.22.16:5060>;tag=289819e859262251i0
Contact: <sip:asterisk@66.192.107.225>
Call-ID: 60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
pbx*CLI>
<-- SIP read from 69.69.144.77:5060:
SIP/2.0 200 OK
To: <sip:jon@192.168.22.16:5060>;tag=289819e859262251i0
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as342ff86c
Call-ID: 60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225
CSeq: 102 INVITE
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK7875712a
Contact: "Jon Webster <3149>" <sip:jon@192.168.22.16:5060>
Server: Sipura/SPA941-4.1.8
Content-Length: 206
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp
v=0
o=- 32922 32922 IN IP4 192.168.22.16
s=-
c=IN IP4 192.168.22.16
t=0 0
m=audio 16392 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--- (11 headers 11 lines)---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.22.16:16392
Peer video RTP is at port 192.168.22.16:65535
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x80006 (gsm|ulaw|h263), 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 <3149> for address/port to send to
Sep 14 22:04:45 WARNING[3730]: chan_sip.c:4029 set_destination: Can't find address for host '3149'
Transmitting (NAT) to 69.69.144.77:5060:
ACK 3149 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK4ff5126d;rport
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as342ff86c
To: <sip:jon@192.168.22.16:5060>;tag=289819e859262251i0
Contact: <sip:asterisk@66.192.107.225>
Call-ID: 60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Retransmitting #4 (NAT) to 69.69.144.77:5060:
BYE 3149 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK30b4d8dc;rport
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as342ff86c
To: <sip:jon@192.168.22.16:5060>;tag=289819e859262251i0
Contact: <sip:asterisk@66.192.107.225>
Call-ID: 60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
pbx*CLI> hangup
No call to hang up
Retransmitting #5 (NAT) to 69.69.144.77:5060:
BYE 3149 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK30b4d8dc;rport
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as342ff86c
To: <sip:jon@192.168.22.16:5060>;tag=289819e859262251i0
Contact: <sip:asterisk@66.192.107.225>
Call-ID: 60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
pbx*CLI>
<-- SIP read from 69.69.144.77:5060:
NOTIFY sip:pbx.elephantoutlook.com SIP/2.0
Via: SIP/2.0/UDP 192.168.22.16:5060;branch=z9hG4bK-876ba3da
From: "Jon Webster <3149>" <sip:jon@pbx.elephantoutlook.com>;tag=5de79c897e369923o0
To: <sip:pbx.elephantoutlook.com>
Call-ID: 37a70dc9-807bd8e3@192.168.22.16
CSeq: 23 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA941-4.1.8
Content-Length: 0
--- (10 headers 0 lines)---
Transmitting (no NAT) to 69.69.144.77:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.22.16:5060;branch=z9hG4bK-876ba3da;received=69.69.144.77
From: "Jon Webster <3149>" <sip:jon@pbx.elephantoutlook.com>;tag=5de79c897e369923o0
To: <sip:pbx.elephantoutlook.com>;tag=as233cc24f
Call-ID: 37a70dc9-807bd8e3@192.168.22.16
CSeq: 23 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Destroying call '37a70dc9-807bd8e3@192.168.22.16'
Retransmitting #6 (NAT) to 69.69.144.77:5060:
BYE 3149 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK30b4d8dc;rport
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as342ff86c
To: <sip:jon@192.168.22.16:5060>;tag=289819e859262251i0
Contact: <sip:asterisk@66.192.107.225>
Call-ID: 60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Sep 14 22:04:54 WARNING[3730]: chan_sip.c:1226 retrans_pkt: Maximum retries exceeded on transmission 60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225 for seqno 103 (Non-critical Request)
Destroying call '60a57ecc4756c4ac16c49ba02e4be7f2@66.192.107.225'
12 headers, 0 lines
Reliably Transmitting (NAT) to 69.69.144.77:5060:
OPTIONS sip:jon@192.168.22.16:5060 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK61b5edb8;rport
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as35ff33bf
To: <sip:jon@192.168.22.16:5060>
Contact: <sip:asterisk@66.192.107.225>
Call-ID: 361a25e26898d0d34b766d16011ea1fe@66.192.107.225
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 15 Sep 2006 02:05:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
pbx*CLI>
<-- SIP read from 69.69.144.77:5060:
SIP/2.0 200 OK
To: <sip:jon@192.168.22.16:5060>;tag=cc407501a462db0bi0
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as35ff33bf
Call-ID: 361a25e26898d0d34b766d16011ea1fe@66.192.107.225
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK61b5edb8
Server: Sipura/SPA941-4.1.8
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
--- (9 headers 0 lines)---
Destroying call '361a25e26898d0d34b766d16011ea1fe@66.192.107.225'
pbx*CLI>
<-- SIP read from 69.69.144.77:5060:
NOTIFY sip:pbx.elephantoutlook.com SIP/2.0
Via: SIP/2.0/UDP 192.168.22.16:5060;branch=z9hG4bK-5bc75d69
From: "Jon Webster <3149>" <sip:jon@pbx.elephantoutlook.com>;tag=5de79c897e369923o0
To: <sip:pbx.elephantoutlook.com>
Call-ID: 37a70dc9-807bd8e3@192.168.22.16
CSeq: 24 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA941-4.1.8
Content-Length: 0
--- (10 headers 0 lines)---
Transmitting (no NAT) to 69.69.144.77:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.22.16:5060;branch=z9hG4bK-5bc75d69;received=69.69.144.77
From: "Jon Webster <3149>" <sip:jon@pbx.elephantoutlook.com>;tag=5de79c897e369923o0
To: <sip:pbx.elephantoutlook.com>;tag=as648fe591
Call-ID: 37a70dc9-807bd8e3@192.168.22.16
CSeq: 24 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Destroying call '37a70dc9-807bd8e3@192.168.22.16'
pbx*CLI>
<-- SIP read from 69.69.144.77:5060:
NOTIFY sip:pbx.elephantoutlook.com SIP/2.0
Via: SIP/2.0/UDP 192.168.22.16:5060;branch=z9hG4bK-e9ff6142
From: "Jon Webster <3149>" <sip:jon@pbx.elephantoutlook.com>;tag=5de79c897e369923o0
To: <sip:pbx.elephantoutlook.com>
Call-ID: 37a70dc9-807bd8e3@192.168.22.16
CSeq: 25 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA941-4.1.8
Content-Length: 0
--- (10 headers 0 lines)---
Transmitting (no NAT) to 69.69.144.77:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.22.16:5060;branch=z9hG4bK-e9ff6142;received=69.69.144.77
From: "Jon Webster <3149>" <sip:jon@pbx.elephantoutlook.com>;tag=5de79c897e369923o0
To: <sip:pbx.elephantoutlook.com>;tag=as5d86d9b4
Call-ID: 37a70dc9-807bd8e3@192.168.22.16
CSeq: 25 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Destroying call '37a70dc9-807bd8e3@192.168.22.16'
pbx*CLI>
<-- SIP read from 69.69.144.77:5060:
NOTIFY sip:pbx.elephantoutlook.com SIP/2.0
Via: SIP/2.0/UDP 192.168.22.16:5060;branch=z9hG4bK-90630d23
From: "Jon Webster <3149>" <sip:jon@pbx.elephantoutlook.com>;tag=5de79c897e369923o0
To: <sip:pbx.elephantoutlook.com>
Call-ID: 37a70dc9-807bd8e3@192.168.22.16
CSeq: 26 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA941-4.1.8
Content-Length: 0
--- (10 headers 0 lines)---
Transmitting (no NAT) to 69.69.144.77:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.22.16:5060;branch=z9hG4bK-90630d23;received=69.69.144.77
From: "Jon Webster <3149>" <sip:jon@pbx.elephantoutlook.com>;tag=5de79c897e369923o0
To: <sip:pbx.elephantoutlook.com>;tag=as62b6fd17
Call-ID: 37a70dc9-807bd8e3@192.168.22.16
CSeq: 26 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Destroying call '37a70dc9-807bd8e3@192.168.22.16'
pbx*CLI>
<-- SIP read from 69.69.144.77:5060:
NOTIFY sip:pbx.elephantoutlook.com SIP/2.0
Via: SIP/2.0/UDP 192.168.22.16:5060;branch=z9hG4bK-3dabf09c
From: "Jon Webster <3149>" <sip:jon@pbx.elephantoutlook.com>;tag=5de79c897e369923o0
To: <sip:pbx.elephantoutlook.com>
Call-ID: 37a70dc9-807bd8e3@192.168.22.16
CSeq: 27 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA941-4.1.8
Content-Length: 0
--- (10 headers 0 lines)---
Transmitting (no NAT) to 69.69.144.77:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.22.16:5060;branch=z9hG4bK-3dabf09c;received=69.69.144.77
From: "Jon Webster <3149>" <sip:jon@pbx.elephantoutlook.com>;tag=5de79c897e369923o0
To: <sip:pbx.elephantoutlook.com>;tag=as46c23d99
Call-ID: 37a70dc9-807bd8e3@192.168.22.16
CSeq: 27 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Destroying call '37a70dc9-807bd8e3@192.168.22.16'
12 headers, 0 lines
Reliably Transmitting (NAT) to 69.69.144.77:5060:
OPTIONS sip:jon@192.168.22.16:5060 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK3f3ba87a;rport
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as2d8fc518
To: <sip:jon@192.168.22.16:5060>
Contact: <sip:asterisk@66.192.107.225>
Call-ID: 2097605a38a33105088262fc7d044cbf@66.192.107.225
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 15 Sep 2006 02:06:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
pbx*CLI>
<-- SIP read from 69.69.144.77:5060:
SIP/2.0 200 OK
To: <sip:jon@192.168.22.16:5060>;tag=cc407501a462db0bi0
From: "asterisk" <sip:asterisk@66.192.107.225>;tag=as2d8fc518
Call-ID: 2097605a38a33105088262fc7d044cbf@66.192.107.225
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK3f3ba87a
Server: Sipura/SPA941-4.1.8
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
--- (9 headers 0 lines)---
Destroying call '2097605a38a33105088262fc7d044cbf@66.192.107.225'
pbx*CLI>
<-- SIP read from 69.69.144.77:5060:
NOTIFY sip:pbx.elephantoutlook.com SIP/2.0
Via: SIP/2.0/UDP 192.168.22.16:5060;branch=z9hG4bK-355d3afc
From: "Jon Webster <3149>" <sip:jon@pbx.elephantoutlook.com>;tag=5de79c897e369923o0
To: <sip:pbx.elephantoutlook.com>
Call-ID: 37a70dc9-807bd8e3@192.168.22.16
CSeq: 28 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA941-4.1.8
Content-Length: 0
--- (10 headers 0 lines)---
Transmitting (no NAT) to 69.69.144.77:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.22.16:5060;branch=z9hG4bK-355d3afc;received=69.69.144.77
From: "Jon Webster <3149>" <sip:jon@pbx.elephantoutlook.com>;tag=5de79c897e369923o0
To: <sip:pbx.elephantoutlook.com>;tag=as3732f817
Call-ID: 37a70dc9-807bd8e3@192.168.22.16
CSeq: 28 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Destroying call '37a70dc9-807bd8e3@192.168.22.16'
pbx*CLI>
<-- SIP read from 69.69.144.77:5060:
NOTIFY sip:pbx.elephantoutlook.com SIP/2.0
Via: SIP/2.0/UDP 192.168.22.16:5060;branch=z9hG4bK-13238063
From: "Jon Webster <3149>" <sip:jon@pbx.elephantoutlook.com>;tag=5de79c897e369923o0
To: <sip:pbx.elephantoutlook.com>
Call-ID: 37a70dc9-807bd8e3@192.168.22.16
CSeq: 29 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Sipura/SPA941-4.1.8
Content-Length: 0
--- (10 headers 0 lines)---
Transmitting (no NAT) to 69.69.144.77:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.22.16:5060;branch=z9hG4bK-13238063;received=69.69.144.77
From: "Jon Webster <3149>" <sip:jon@pbx.elephantoutlook.com>;tag=5de79c897e369923o0
To: <sip:pbx.elephantoutlook.com>;tag=as2246e224
Call-ID: 37a70dc9-807bd8e3@192.168.22.16
CSeq: 29 NOTIFY
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Destroying call '37a70dc9-807bd8e3@192.168.22.16'
pbx*CLI>