All pastes #270319 Raw Edit

Anonymous

public text v1 · immutable
·published 2006-12-07 07:29 UTC
rendered paste body
    -- Executing NoOp("SIP/jon-0817d508", "Exchange Test: 666666") in new stack
    -- Executing Dial("SIP/jon-0817d508", "SIP/pbx.elephantoutlook.com:5070/666666") in new stack
We're at 66.192.107.225 port 13976
Video is at 66.192.107.225 port 14142
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 (no NAT) to 66.192.107.225:5070:
INVITE sip:666666@pbx.elephantoutlook.com:5070 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK313e9ff7;rport
From: "Jon Webster" <sip:3149@66.192.107.225>;tag=as5c3450e3
To: <sip:666666@pbx.elephantoutlook.com:5070>
Contact: <sip:3149@66.192.107.225>
Call-ID: 4d1f33b47f8cbc2237b44eb051f1f17e@66.192.107.225
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 07 Dec 2006 07:19:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 291

v=0
o=root 8640 8640 IN IP4 66.192.107.225
s=session
c=IN IP4 66.192.107.225
t=0 0
m=audio 13976 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 14142 RTP/AVP 34
a=rtpmap:34 H263/90000

---
    -- Called pbx.elephantoutlook.com:5070/666666
pbx*CLI>
<-- SIP read from 66.192.107.225:5070:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK313e9ff7;rport=5060
From: "Jon Webster" <sip:3149@66.192.107.225>;tag=as5c3450e3
To: <sip:666666@pbx.elephantoutlook.com:5070>
Call-ID: 4d1f33b47f8cbc2237b44eb051f1f17e@66.192.107.225
CSeq: 102 INVITE
Server: OpenSer (1.1.0-notls (i386/linux))
Content-Length: 0
Warning: 392 66.192.107.225:5070 "Noisy feedback tells:  pid=20826 req_src_ip=66.192.107.225 req_src_port=5060 in_uri=sip:666666@pbx.elephantoutlook.com:5070 out_uri=sip:666666@pbx.elephantoutlook.com:5070 via_cnt==1"


--- (9 headers 0 lines)---
pbx*CLI>
<-- SIP read from 66.192.107.225:5070:
SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL)
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK313e9ff7;rport=5060
From: "Jon Webster" <sip:3149@66.192.107.225>;tag=as5c3450e3
To: <sip:666666@pbx.elephantoutlook.com:5070>;tag=ff0593a3ce80005043f79ea67dccae1c.3701
Call-ID: 4d1f33b47f8cbc2237b44eb051f1f17e@66.192.107.225
CSeq: 102 INVITE
Server: OpenSer (1.1.0-notls (i386/linux))
Content-Length: 0
Warning: 392 66.192.107.225:5070 "Noisy feedback tells:  pid=20826 req_src_ip=66.192.107.225 req_src_port=5060 in_uri=sip:666666@pbx.elephantoutlook.com:5070 out_uri=sip:666666@pbx.elephantoutlook.com:5070 via_cnt==1"


--- (9 headers 0 lines)---
    -- Got SIP response 500 "I'm terribly sorry, server error occurred (1/SL)" back from 66.192.107.225
Transmitting (no NAT) to 66.192.107.225:5070:
ACK sip:666666@pbx.elephantoutlook.com:5070 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK313e9ff7;rport
From: "Jon Webster" <sip:3149@66.192.107.225>;tag=as5c3450e3
To: <sip:666666@pbx.elephantoutlook.com:5070>;tag=ff0593a3ce80005043f79ea67dccae1c.3701
Contact: <sip:3149@66.192.107.225>
Call-ID: 4d1f33b47f8cbc2237b44eb051f1f17e@66.192.107.225
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/pbx.elephantoutlook.com:5070-0823d1a0 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/jon-0817d508' status is 'CONGESTION'
pbx*CLI>
<-- SIP read from 66.192.107.225:5070:
SIP/2.0 302 Moved Temporarily
FROM: "Jon Webster"<sip:3149@66.192.107.225>;tag=as5c3450e3
TO: <sip:666666@pbx.elephantoutlook.com:5070>;tag=3c2de362d0
CSEQ: 102 INVITE
CALL-ID: 4d1f33b47f8cbc2237b44eb051f1f17e@66.192.107.225
MAX-FORWARDS: 70
VIA: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK313e9ff7;rport=5060
CONTENT-LENGTH: 0
SERVER: RTCC/2.0.6010.0
CONTACT: <sip:666666@pbx.elephantoutlook.com:5061;transport=TCP>


--- (10 headers 0 lines)---
    -- Got SIP response 302 "Moved Temporarily" back from 66.192.107.225
Transmitting (no NAT) to 66.192.107.225:5070:
ACK sip:666666@pbx.elephantoutlook.com:5070 SIP/2.0
Via: SIP/2.0/UDP 66.192.107.225:5060;branch=z9hG4bK313e9ff7;rport
From: "Jon Webster" <sip:3149@66.192.107.225>;tag=as5c3450e3
To: <sip:666666@pbx.elephantoutlook.com:5070>;tag=3c2de362d0
Contact: <sip:3149@66.192.107.225>
Call-ID: 4d1f33b47f8cbc2237b44eb051f1f17e@66.192.107.225
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Destroying call '4d1f33b47f8cbc2237b44eb051f1f17e@66.192.107.225'
pbx*CLI>
<-- SIP read from 66.192.107.225:5070:
OPTIONS sip:pbx.elephantoutlook.com:5060;transport=TCP SIP/2.0
Record-Route: <sip:66.192.107.225:5070;transport=tcp;lr=on;ftag=1ed053927e>
FROM: <sip:eo-dev.webcode.com:5060>;epid=8C-D4-0D-19-22;tag=1ed053927e
TO: <sip:pbx.elephantoutlook.com:5060;transport=TCP>
CSEQ: 312 OPTIONS
CALL-ID: 2b0405658d76427999f7dec5c813028b
MAX-FORWARDS: 69
Via: SIP/2.0/UDP 66.192.107.225:5070;branch=z9hG4bKc87c.cfc2eb.0;i=9
VIA: SIP/2.0/TCP 66.192.106.194:3437;branch=z9hG4bKffb45234
ACCEPT: application/sdp
CONTACT: <sip:eo-dev.webcode.com:5060;transport=Tcp;maddr=66.192.106.194>
CONTENT-LENGTH: 0
USER-AGENT: RTCC/2.0.6010.0


--- (13 headers 0 lines)---
Looking for s in local (domain pbx.elephantoutlook.com)
Transmitting (no NAT) to 66.192.107.225:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.192.107.225:5070;branch=z9hG4bKc87c.cfc2eb.0;i=9;received=66.192.107.225
Via: SIP/2.0/TCP 66.192.106.194:3437;branch=z9hG4bKffb45234
Record-Route: <sip:66.192.107.225:5070;transport=tcp;lr=on;ftag=1ed053927e>
From: <sip:eo-dev.webcode.com:5060>;epid=8C-D4-0D-19-22;tag=1ed053927e
To: <sip:pbx.elephantoutlook.com:5060;transport=TCP>;tag=as5f48131a
Call-ID: 2b0405658d76427999f7dec5c813028b
CSeq: 312 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:66.192.107.225>
Accept: application/sdp
Content-Length: 0


---
Destroying call '2b0405658d76427999f7dec5c813028b'
pbx*CLI>
<-- SIP read from 66.192.107.225:5070:
OPTIONS sip:pbx.elephantoutlook.com:5060;transport=TCP SIP/2.0
Record-Route: <sip:66.192.107.225:5070;transport=tcp;lr=on;ftag=7e9f7b41ce>
FROM: <sip:eo-dev.webcode.com:5060>;epid=8C-D4-0D-19-22;tag=7e9f7b41ce
TO: <sip:pbx.elephantoutlook.com:5060;transport=TCP>
CSEQ: 313 OPTIONS
CALL-ID: b63c8e8c7d6c40f8825c34c689937a10
MAX-FORWARDS: 69
Via: SIP/2.0/UDP 66.192.107.225:5070;branch=z9hG4bK9b25.c3bddd12.0;i=a
VIA: SIP/2.0/TCP 66.192.106.194:3446;branch=z9hG4bK68c390d5
ACCEPT: application/sdp
CONTACT: <sip:eo-dev.webcode.com:5060;transport=Tcp;maddr=66.192.106.194>
CONTENT-LENGTH: 0
USER-AGENT: RTCC/2.0.6010.0


--- (13 headers 0 lines)---
Looking for s in local (domain pbx.elephantoutlook.com)
Transmitting (no NAT) to 66.192.107.225:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.192.107.225:5070;branch=z9hG4bK9b25.c3bddd12.0;i=a;received=66.192.107.225
Via: SIP/2.0/TCP 66.192.106.194:3446;branch=z9hG4bK68c390d5
Record-Route: <sip:66.192.107.225:5070;transport=tcp;lr=on;ftag=7e9f7b41ce>
From: <sip:eo-dev.webcode.com:5060>;epid=8C-D4-0D-19-22;tag=7e9f7b41ce
To: <sip:pbx.elephantoutlook.com:5060;transport=TCP>;tag=as5b3ab11b
Call-ID: b63c8e8c7d6c40f8825c34c689937a10
CSeq: 313 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:66.192.107.225>
Accept: application/sdp
Content-Length: 0


---
Destroying call 'b63c8e8c7d6c40f8825c34c689937a10'
pbx*CLI>
<-- SIP read from 66.192.107.225:5070:
OPTIONS sip:pbx.elephantoutlook.com:5060;transport=TCP SIP/2.0
Record-Route: <sip:66.192.107.225:5070;transport=tcp;lr=on;ftag=8a6030c58c>
FROM: <sip:eo-dev.webcode.com:5060>;epid=8C-D4-0D-19-22;tag=8a6030c58c
TO: <sip:pbx.elephantoutlook.com:5060;transport=TCP>
CSEQ: 314 OPTIONS
CALL-ID: 238060bb462b4b6fbdee9352fb924389
MAX-FORWARDS: 69
Via: SIP/2.0/UDP 66.192.107.225:5070;branch=z9hG4bKf87e.5be96a12.0;i=b
VIA: SIP/2.0/TCP 66.192.106.194:3453;branch=z9hG4bK197a6ad9
ACCEPT: application/sdp
CONTACT: <sip:eo-dev.webcode.com:5060;transport=Tcp;maddr=66.192.106.194>
CONTENT-LENGTH: 0
USER-AGENT: RTCC/2.0.6010.0


--- (13 headers 0 lines)---
Looking for s in local (domain pbx.elephantoutlook.com)
Transmitting (no NAT) to 66.192.107.225:5070:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 66.192.107.225:5070;branch=z9hG4bKf87e.5be96a12.0;i=b;received=66.192.107.225
Via: SIP/2.0/TCP 66.192.106.194:3453;branch=z9hG4bK197a6ad9
Record-Route: <sip:66.192.107.225:5070;transport=tcp;lr=on;ftag=8a6030c58c>
From: <sip:eo-dev.webcode.com:5060>;epid=8C-D4-0D-19-22;tag=8a6030c58c
To: <sip:pbx.elephantoutlook.com:5060;transport=TCP>;tag=as4226570b
Call-ID: 238060bb462b4b6fbdee9352fb924389
CSeq: 314 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:66.192.107.225>
Accept: application/sdp
Content-Length: 0


---
Destroying call '238060bb462b4b6fbdee9352fb924389'
pbx*CLI>