All pastes #407407 Raw Edit

hillct

public text v1 · immutable
#407407 ·published 2007-03-24 00:09 UTC
rendered paste body
---
filegate2*CLI>
<-- SIP read from 216.143.130.36:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK138a7671;rport=5060
From: <sip:hillct@sip.voipyourphone.com>;tag=as1051febd
To: <sip:hillct@sip.voipyourphone.com>;tag=68d238da7d1832978b1a894dbad8a35e.5d9d
Call-ID: 681293625380292600bf2f4f510faa07@192.168.1.76
CSeq: 433 REGISTER
Contact: <sip:s@192.168.1.76:5060>;expires=120
Server: OpenSer (1.0.0 (i386/freebsd))
Content-Length: 0


--- (9 headers 0 lines) ---
Scheduling destruction of call '681293625380292600bf2f4f510faa07@192.168.1.76' in 32000 ms
Mar 23 20:01:55 NOTICE[163848]: chan_sip.c:12279 handle_response_register: Outbound Registration: Expiry for sip.voipyourphone.com is 120 sec (Scheduling reregistration in 105 s)
filegate2*CLI>
<-- SIP read from 216.143.130.36:5060:
NOTIFY sip:s@209.42.200.5:5060 SIP/2.0
Max-Forwards: 10
Record-Route: <sip:216.143.130.36;lr=on>
Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bK4265.16330f61.0
Via: SIP/2.0/UDP resi-sip-01.rnktel.com:38896;received=216.143.130.47;rport=38897
From: "Unknown" <sip:Unknown@216.143.130.36>
To: <sip:hillct@216.143.130.36>
Contact: <sip:Unknown@216.143.130.36>
Call-ID: 718111ed3c1ed2fb3Dx3Ow3Co4To4Bp7@216.143.130.47
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 36

Messages-Waiting: no
Voicemail: 0/0

--- (14 headers 2 lines) ---
Transmitting (NAT) to 216.143.130.36:5060:
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bK4265.16330f61.0;received=216.143.130.36
Via: SIP/2.0/UDP resi-sip-01.rnktel.com:38896;received=216.143.130.47;rport=38897
From: "Unknown" <sip:Unknown@216.143.130.36>
To: <sip:hillct@216.143.130.36>;tag=as46b030cc
Call-ID: 718111ed3c1ed2fb3Dx3Ow3Co4To4Bp7@216.143.130.47
CSeq: 102 NOTIFY
User-Agent: OpenPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Content-Length: 0


---
filegate2*CLI>
<-- SIP read from 192.168.1.147:5060:
NOTIFY sip:192.168.1.76 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK-65ac76ba
From: 6001 <sip:6001@192.168.1.76>;tag=95f215bae45e8b41o0
To: <sip:192.168.1.76>
Call-ID: 6c4ec45e-9ce70ced@192.168.1.147
CSeq: 1587 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Linksys/SPA2102-3.3.6
Content-Length: 0


--- (10 headers 0 lines) ---
Transmitting (NAT) to 192.168.1.147:5060:
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK-65ac76ba;received=192.168.1.147
From: 6001 <sip:6001@192.168.1.76>;tag=95f215bae45e8b41o0
To: <sip:192.168.1.76>;tag=as445c32ac
Call-ID: 6c4ec45e-9ce70ced@192.168.1.147
CSeq: 1587 NOTIFY
User-Agent: OpenPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Content-Length: 0


---
filegate2*CLI>
<-- SIP read from 216.143.130.36:5060:
INVITE sip:s@209.42.200.5:5060 SIP/2.0
Max-Forwards: 10
Record-Route: <sip:216.143.130.36;ftag=as3e769388;lr=on>
Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bKaa66.05c5cd87.0
Via: SIP/2.0/UDP 216.143.130.95:5060;branch=z9hG4bK32b07293;rport=5060
From: "4150050473" <sip:4150050473@216.143.130.95>;tag=as3e769388
To: <sip:hillct@216.143.130.36>
Contact: <sip:4150050473@216.143.130.95>
Call-ID: 21aafb7239a9f4101d6ad3bf4ad51310@216.143.130.95
CSeq: 102 INVITE
ser-Agent: Asterisk PBX
Date: Sat, 24 Mar 2007 00:02:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 319

v=0
o=root 4649 4649 IN IP4 216.143.130.95
s=session
c=IN IP4 216.143.130.95
t=0 0
m=audio 48508 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
m=video 23128 RTP/AVP 31 34
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000

--- (15 headers 14 lines) ---
Using INVITE request as basis request - 21aafb7239a9f4101d6ad3bf4ad51310@216.143.130.95
Sending to 216.143.130.36 : 5060 (NAT)
Found peer 'sip_proxy-out'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Mar 23 20:02:10 WARNING[163848]: chan_sip.c:4704 process_sdp: Unknown SDP media type in offer: video 23128 RTP/AVP 31 34
Peer audio RTP is at port 216.143.130.95:48508
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Found description format H261
Found description format H263
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for s in default (domain 209.42.200.5)
list_route: hop: <sip:216.143.130.36;ftag=as3e769388;lr=on>
Transmitting (NAT) to 216.143.130.36:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bKaa66.05c5cd87.0;received=216.143.130.36
Via: SIP/2.0/UDP 216.143.130.95:5060;branch=z9hG4bK32b07293;rport=5060
From: "4150050473" <sip:4150050473@216.143.130.95>;tag=as3e769388
To: <sip:hillct@216.143.130.36>
Call-ID: 21aafb7239a9f4101d6ad3bf4ad51310@216.143.130.95
CSeq: 102 INVITE
User-Agent: OpenPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:s@192.168.1.76>
Content-Length: 0


---
    -- Executing FaxDetect("SIP/hillct-d3ac", "4000") in new stack
We're at 192.168.1.76 port 12652
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 216.143.130.36:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bKaa66.05c5cd87.0;received=216.143.130.36
Via: SIP/2.0/UDP 216.143.130.95:5060;branch=z9hG4bK32b07293;rport=5060
Record-Route: <sip:216.143.130.36;ftag=as3e769388;lr=on>
rom: "4150050473" <sip:4150050473@216.143.130.95>;tag=as3e769388
To: <sip:hillct@216.143.130.36>;tag=as15686f6a
Call-ID: 21aafb7239a9f4101d6ad3bf4ad51310@216.143.130.95
CSeq: 102 INVITE
User-Agent: OpenPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:s@192.168.1.76>
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 2764 2764 IN IP4 192.168.1.76
s=session
c=IN IP4 192.168.1.76
t=0 0
m=audio 12652 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
filegate2*CLI>
<-- SIP read from 216.143.130.36:5060:
ACK sip:s@209.42.200.5:5060 SIP/2.0
Max-Forwards: 10
Record-Route: <sip:216.143.130.36;ftag=as3e769388;lr=on>
Via: SIP/2.0/UDP 216.143.130.36;branch=0
Via: SIP/2.0/UDP 216.143.130.95:5060;branch=z9hG4bK08a58ec3;rport=5060
From: "4150050473" <sip:4150050473@216.143.130.95>;tag=as3e769388
To: <sip:hillct@216.143.130.36>;tag=as15686f6a
Contact: <sip:4150050473@216.143.130.95>
Call-ID: 21aafb7239a9f4101d6ad3bf4ad51310@216.143.130.95
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0


--- (12 headers 0 lines) ---
Mar 23 20:02:11 NOTICE[1441805]: app_faxdetect.c:306 detectfax_exec: Redirecting SIP/hillct-d3ac to fax extension [DTMF]
    -- Executing Dial("SIP/hillct-d3ac", "SIP/5000") in new stack
We're at 192.168.1.76 port 13696
Adding codec 0x4 (ulaw) to SDP
13 headers, 8 lines
Reliably Transmitting (NAT) to 192.168.1.147:5060:
INVITE sip:5000@192.168.1.147:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK23175690;rport
From: "4150050473" <sip:4150050473@192.168.1.76>;tag=as72d92084
To: <sip:5000@192.168.1.147:5060>
Contact: <sip:4150050473@192.168.1.76>
Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76
CSeq: 102 INVITE
User-Agent: OpenPBX
Max-Forwards: 70
Date: Sat, 24 Mar 2007 00:02:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 158

v=0
o=root 2764 2764 IN IP4 192.168.1.76
s=session
c=IN IP4 192.168.1.76
t=0 0
m=audio 13696 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

---
    -- Called 5000
Mar 23 20:02:11 NOTICE[1441805]: channel.c:1627 opbx_read: Dropping incompatible voice frame on SIP/hillct-d3ac of format slin since our native format has changed to ulaw
filegate2*CLI>
<-- SIP read from 192.168.1.147:5060:
SIP/2.0 100 Trying
To: <sip:5000@192.168.1.147:5060>
From: "4150050473" <sip:4150050473@192.168.1.76>;tag=as72d92084
Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK23175690
Server: Linksys/SPA2102-3.3.6
Content-Length: 0


--- (8 headers 0 lines) ---
filegate2*CLI>
<-- SIP read from 192.168.1.147:5060:
SIP/2.0 180 Ringing
To: <sip:5000@192.168.1.147:5060>;tag=cca40ac71da83e1ei1
From: "4150050473" <sip:4150050473@192.168.1.76>;tag=as72d92084
Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK23175690
Server: Linksys/SPA2102-3.3.6
Remote-Party-ID: 5000 <sip:5000@192.168.1.76>;screen=yes;party=called
Content-Length: 0


--- (9 headers 0 lines) ---
    -- SIP/5000-b6c5 is ringing
filegate2*CLI>
<-- SIP read from 192.168.1.147:5060:
NOTIFY sip:192.168.1.76 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK-9fd3eaec
From: 6001 <sip:6001@192.168.1.76>;tag=95f215bae45e8b41o0
To: <sip:192.168.1.76>
Call-ID: 6c4ec45e-9ce70ced@192.168.1.147
CSeq: 1588 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Linksys/SPA2102-3.3.6
Content-Length: 0


--- (10 headers 0 lines) ---
Transmitting (NAT) to 192.168.1.147:5060:
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK-9fd3eaec;received=192.168.1.147
From: 6001 <sip:6001@192.168.1.76>;tag=95f215bae45e8b41o0
To: <sip:192.168.1.76>;tag=as4ddcdd9e
Call-ID: 6c4ec45e-9ce70ced@192.168.1.147
CSeq: 1588 NOTIFY
User-Agent: OpenPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Content-Length: 0


---
filegate2*CLI>
<-- SIP read from 216.143.130.36:5060:
NOTIFY sip:s@209.42.200.5:5060 SIP/2.0
Max-Forwards: 10
Record-Route: <sip:216.143.130.36;lr=on>
Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bKcbfc.4d1d527.0
Via: SIP/2.0/UDP resi-sip-01.rnktel.com:39302;received=216.143.130.47;rport=39303
From: "Unknown" <sip:Unknown@216.143.130.36>
To: <sip:hillct@216.143.130.36>
Contact: <sip:Unknown@216.143.130.36>
Call-ID: 718111ed3c1ed2fb3Lx3Tw5Ml5Op1Ax6@216.143.130.47
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 36

Messages-Waiting: no
Voicemail: 0/0

--- (14 headers 2 lines) ---
Transmitting (NAT) to 216.143.130.36:5060:
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bKcbfc.4d1d527.0;received=216.143.130.36
Via: SIP/2.0/UDP resi-sip-01.rnktel.com:39302;received=216.143.130.47;rport=39303
From: "Unknown" <sip:Unknown@216.143.130.36>
To: <sip:hillct@216.143.130.36>;tag=as1ae6b4d1
Call-ID: 718111ed3c1ed2fb3Lx3Tw5Ml5Op1Ax6@216.143.130.47
CSeq: 102 NOTIFY
User-Agent: OpenPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Content-Length: 0


---
filegate2*CLI>
<-- SIP read from 192.168.1.147:5060:
NOTIFY sip:192.168.1.76 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK-7906da40
From: 6001 <sip:6001@192.168.1.76>;tag=95f215bae45e8b41o0
To: <sip:192.168.1.76>
Call-ID: 6c4ec45e-9ce70ced@192.168.1.147
CSeq: 1589 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Linksys/SPA2102-3.3.6
Content-Length: 0


--- (10 headers 0 lines) ---
Transmitting (NAT) to 192.168.1.147:5060:
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK-7906da40;received=192.168.1.147
From: 6001 <sip:6001@192.168.1.76>;tag=95f215bae45e8b41o0
To: <sip:192.168.1.76>;tag=as6d953122
Call-ID: 6c4ec45e-9ce70ced@192.168.1.147
CSeq: 1589 NOTIFY
User-Agent: OpenPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Content-Length: 0


---
Destroying call '681293625380292600bf2f4f510faa07@192.168.1.76'
filegate2*CLI>
<-- SIP read from 216.143.130.36:5060:

--- (0 headers 0 lines) Nat keepalive ---
filegate2*CLI>
<-- SIP read from 192.168.1.147:5060:
SIP/2.0 200 OK
To: <sip:5000@192.168.1.147:5060>;tag=cca40ac71da83e1ei1
From: "4150050473" <sip:4150050473@192.168.1.76>;tag=as72d92084
Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK23175690
Contact: 5000 <sip:5000@192.168.1.147:5060>
Server: Linksys/SPA2102-3.3.6
Remote-Party-ID: 5000 <sip:5000@192.168.1.76>;screen=yes;party=called
Content-Length: 257
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 2387898 2387898 IN IP4 192.168.1.147
s=-
c=IN IP4 192.168.1.147
t=0 0
m=audio 16390 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

--- (13 headers 13 lines) ---
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.147:16390
Found description format PCMU
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), 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:5000@192.168.1.147:5060>
set_destination: Parsing <sip:5000@192.168.1.147:5060> for address/port to send to
set_destination: set destination to 192.168.1.147, port 5060
Transmitting (NAT) to 192.168.1.147:5060:
ACK sip:5000@192.168.1.147:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK42b6fc9e;rport
From: "4150050473" <sip:4150050473@192.168.1.76>;tag=as72d92084
To: <sip:5000@192.168.1.147:5060>;tag=cca40ac71da83e1ei1
Contact: <sip:4150050473@192.168.1.76>
Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76
CSeq: 102 ACK
User-Agent: OpenPBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/5000-b6c5 answered SIP/hillct-d3ac
    -- Attempting native bridge of SIP/hillct-d3ac and SIP/5000-b6c5
set_destination: Parsing <sip:216.143.130.36;ftag=as3e769388;lr=on> for address/port to send to
set_destination: set destination to 216.143.130.36, port 5060
We're at 192.168.1.76 port 12652
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
14 headers, 10 lines
Reliably Transmitting (NAT) to 216.143.130.36:5060:
INVITE sip:4150050473@216.143.130.95 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK05fc6954;rport
Route: <sip:216.143.130.36;ftag=as3e769388;lr=on>
From: <sip:hillct@216.143.130.36>;tag=as15686f6a
To: "4150050473" <sip:4150050473@216.143.130.95>;tag=as3e769388
Contact: <sip:s@192.168.1.76>
Call-ID: 21aafb7239a9f4101d6ad3bf4ad51310@216.143.130.95
CSeq: 102 INVITE
User-Agent: OpenPBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-openpbx-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 216

v=0
o=root 2764 2765 IN IP4 192.168.1.147
s=session
c=IN IP4 192.168.1.147
t=0 0
m=audio 16390 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
set_destination: Parsing <sip:5000@192.168.1.147:5060> for address/port to send to
set_destination: set destination to 192.168.1.147, port 5060
We're at 192.168.1.76 port 13696
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (NAT) to 192.168.1.147:5060:
INVITE sip:5000@192.168.1.147:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK09e9244d;rport
From: "4150050473" <sip:4150050473@192.168.1.76>;tag=as72d92084
To: <sip:5000@192.168.1.147:5060>;tag=cca40ac71da83e1ei1
Contact: <sip:4150050473@192.168.1.76>
Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76
CSeq: 103 INVITE
User-Agent: OpenPBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-openpbx-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 2764 2765 IN IP4 216.143.130.95
s=session
c=IN IP4 216.143.130.95
t=0 0
m=audio 48508 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
filegate2*CLI>
<-- SIP read from 192.168.1.147:5060:
SIP/2.0 200 OK
To: <sip:5000@192.168.1.147:5060>;tag=cca40ac71da83e1ei1
From: "4150050473" <sip:4150050473@192.168.1.76>;tag=as72d92084
Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76
CSeq: 103 INVITE
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK09e9244d
Contact: 5000 <sip:5000@192.168.1.147:5060>
Server: Linksys/SPA2102-3.3.6
Content-Length: 284
Content-Type: application/sdp

v=0
o=- 2389749 2389749 IN IP4 192.168.1.147
s=-
c=IN IP4 192.168.1.147
t=0 0
m=audio 16390 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -

--- (10 headers 14 lines) ---
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.147:16390
Found description format PCMU
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), 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)
Mar 23 20:02:30 WARNING[163848]: chan_sip.c:12037 handle_response_invite: Strange... The other side of the bridge don't have udptl struct
set_destination: Parsing <sip:5000@192.168.1.147:5060> for address/port to send to
set_destination: set destination to 192.168.1.147, port 5060
Transmitting (NAT) to 192.168.1.147:5060:
ACK sip:5000@192.168.1.147:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK588b5634;rport
From: "4150050473" <sip:4150050473@192.168.1.76>;tag=as72d92084
To: <sip:5000@192.168.1.147:5060>;tag=cca40ac71da83e1ei1
Contact: <sip:4150050473@192.168.1.76>
Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76
CSeq: 103 ACK
User-Agent: OpenPBX
Max-Forwards: 70
Content-Length: 0


---
filegate2*CLI>
<-- SIP read from 216.143.130.36:5060:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 209.42.200.5:5060;branch=z9hG4bK05fc6954;rport=5060
From: <sip:hillct@216.143.130.36>;tag=as15686f6a
To: "4150050473" <sip:4150050473@216.143.130.95>;tag=as3e769388
Call-ID: 21aafb7239a9f4101d6ad3bf4ad51310@216.143.130.95
CSeq: 102 INVITE
Server: OpenSer (1.0.0 (i386/freebsd))
Content-Length: 0


--- (8 headers 0 lines) ---
filegate2*CLI>
<-- SIP read from 216.143.130.36:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.42.200.5:5060;branch=z9hG4bK05fc6954;rport=5060
Record-Route: <sip:216.143.130.36;ftag=as15686f6a;lr=on>
From: <sip:hillct@216.143.130.36>;tag=as15686f6a
To: "4150050473" <sip:4150050473@216.143.130.95>;tag=as3e769388
Call-ID: 21aafb7239a9f4101d6ad3bf4ad51310@216.143.130.95
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:4150050473@216.143.130.95:5060>
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 4649 4650 IN IP4 216.143.130.95
s=session
c=IN IP4 216.143.130.95
t=0 0
m=audio 48508 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=direction:active

--- (12 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 216.143.130.95:48508
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|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: <sip:216.143.130.36;ftag=as15686f6a;lr=on>
Mar 23 20:02:30 WARNING[163848]: chan_sip.c:12037 handle_response_invite: Strange... The other side of the bridge don't have udptl struct
set_destination: Parsing <sip:216.143.130.36;ftag=as15686f6a;lr=on> for address/port to send to
set_destination: set destination to 216.143.130.36, port 5060
Transmitting (NAT) to 216.143.130.36:5060:
ACK sip:4150050473@216.143.130.95:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK231e5826;rport
Route: <sip:216.143.130.36;ftag=as15686f6a;lr=on>
From: <sip:hillct@216.143.130.36>;tag=as15686f6a
To: "4150050473" <sip:4150050473@216.143.130.95>;tag=as3e769388
Contact: <sip:s@192.168.1.76>
Call-ID: 21aafb7239a9f4101d6ad3bf4ad51310@216.143.130.95
CSeq: 102 ACK
User-Agent: OpenPBX
Max-Forwards: 70
Content-Length: 0


---
filegate2*CLI>
<-- SIP read from 216.143.130.36:5060:
BYE sip:s@209.42.200.5:5060 SIP/2.0
Max-Forwards: 10
Record-Route: <sip:216.143.130.36;ftag=as3e769388;lr=on>
Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bKba66.697031b4.0
Via: SIP/2.0/UDP 216.143.130.95:5060;branch=z9hG4bK5737f314;rport=5060
From: "4150050473" <sip:4150050473@216.143.130.95>;tag=as3e769388
To: <sip:hillct@216.143.130.36>;tag=as15686f6a
Contact: <sip:4150050473@216.143.130.95>
Call-ID: 21aafb7239a9f4101d6ad3bf4ad51310@216.143.130.95
CSeq: 103 BYE
User-Agent: Asterisk PBX
Content-Length: 0


--- (12 headers 0 lines) ---
Sending to 216.143.130.36 : 5060 (NAT)
Transmitting (NAT) to 216.143.130.36:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bKba66.697031b4.0;received=216.143.130.36
Via: SIP/2.0/UDP 216.143.130.95:5060;branch=z9hG4bK5737f314;rport=5060
Record-Route: <sip:216.143.130.36;ftag=as3e769388;lr=on>
From: "4150050473" <sip:4150050473@216.143.130.95>;tag=as3e769388
To: <sip:hillct@216.143.130.36>;tag=as15686f6a
Call-ID: 21aafb7239a9f4101d6ad3bf4ad51310@216.143.130.95
CSeq: 103 BYE
User-Agent: OpenPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:s@192.168.1.76>
Content-Length: 0
X-OpenPBX-HangupCause: Normal Clearing


---
set_destination: Parsing <sip:5000@192.168.1.147:5060> for address/port to send to
set_destination: set destination to 192.168.1.147, port 5060
We're at 192.168.1.76 port 13696
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (NAT) to 192.168.1.147:5060:
INVITE sip:5000@192.168.1.147:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK67e7b98c;rport
From: "4150050473" <sip:4150050473@192.168.1.76>;tag=as72d92084
To: <sip:5000@192.168.1.147:5060>;tag=cca40ac71da83e1ei1
Contact: <sip:4150050473@192.168.1.76>
Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76
CSeq: 104 INVITE
User-Agent: OpenPBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-openpbx-info: SIP re-invite (RTP bridge)
Content-Type: application/sdp
Content-Length: 214

v=0
o=root 2764 2766 IN IP4 192.168.1.76
s=session
c=IN IP4 192.168.1.76
t=0 0
m=audio 13696 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
  == Spawn extension (default, fax, 1) exited non-zero on 'SIP/hillct-d3ac'
filegate2*CLI>
<-- SIP read from 192.168.1.147:5060:
SIP/2.0 200 OK
To: <sip:5000@192.168.1.147:5060>;tag=cca40ac71da83e1ei1
From: "4150050473" <sip:4150050473@192.168.1.76>;tag=as72d92084
Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76
CSeq: 104 INVITE
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK67e7b98c
Contact: 5000 <sip:5000@192.168.1.147:5060>
Server: Linksys/SPA2102-3.3.6
Content-Length: 284
Content-Type: application/sdp

v=0
o=- 2390630 2390630 IN IP4 192.168.1.147
s=-
c=IN IP4 192.168.1.147
t=0 0
m=audio 16390 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -

--- (10 headers 14 lines) ---
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.147:16390
Found description format PCMU
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), 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:5000@192.168.1.147:5060> for address/port to send to
set_destination: set destination to 192.168.1.147, port 5060
Transmitting (NAT) to 192.168.1.147:5060:
ACK sip:5000@192.168.1.147:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK482af03f;rport
From: "4150050473" <sip:4150050473@192.168.1.76>;tag=as72d92084
To: <sip:5000@192.168.1.147:5060>;tag=cca40ac71da83e1ei1
Contact: <sip:4150050473@192.168.1.76>
Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76
CSeq: 104 ACK
User-Agent: OpenPBX
Max-Forwards: 70
Content-Length: 0


---
set_destination: Parsing <sip:5000@192.168.1.147:5060> for address/port to send to
set_destination: set destination to 192.168.1.147, port 5060
Reliably Transmitting (NAT) to 192.168.1.147:5060:
BYE sip:5000@192.168.1.147:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK20e56adf;rport
From: "4150050473" <sip:4150050473@192.168.1.76>;tag=as72d92084
To: <sip:5000@192.168.1.147:5060>;tag=cca40ac71da83e1ei1
Contact: <sip:4150050473@192.168.1.76>
Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76
CSeq: 105 BYE
User-Agent: OpenPBX
Max-Forwards: 70
Content-Length: 0


---

<-- SIP read from 192.168.1.147:5060:
SIP/2.0 200 OK
To: <sip:5000@192.168.1.147:5060>;tag=cca40ac71da83e1ei1
From: "4150050473" <sip:4150050473@192.168.1.76>;tag=as72d92084
Call-ID: 4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76
CSeq: 105 BYE
Via: SIP/2.0/UDP 192.168.1.76:5060;branch=z9hG4bK20e56adf
Server: Linksys/SPA2102-3.3.6
Content-Length: 0


--- (8 headers 0 lines) ---
Destroying call '4f984df71b4ccfdf3fa56249233e7e6a@192.168.1.76'
Destroying call '21aafb7239a9f4101d6ad3bf4ad51310@216.143.130.95'
filegate2*CLI>
<-- SIP read from 192.168.1.147:5060:
NOTIFY sip:192.168.1.76 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK-88a698d9
From: 6001 <sip:6001@192.168.1.76>;tag=95f215bae45e8b41o0
To: <sip:192.168.1.76>
Call-ID: 6c4ec45e-9ce70ced@192.168.1.147
CSeq: 1590 NOTIFY
Max-Forwards: 70
Event: keep-alive
User-Agent: Linksys/SPA2102-3.3.6
Content-Length: 0


--- (10 headers 0 lines) ---
Transmitting (NAT) to 192.168.1.147:5060:
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 192.168.1.147:5060;branch=z9hG4bK-88a698d9;received=192.168.1.147
From: 6001 <sip:6001@192.168.1.76>;tag=95f215bae45e8b41o0
To: <sip:192.168.1.76>;tag=as096499c1
Call-ID: 6c4ec45e-9ce70ced@192.168.1.147
CSeq: 1590 NOTIFY
User-Agent: OpenPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Content-Length: 0


---
filegate2*CLI>