All pastes #674252 Raw Edit

Miscellany

public text v1 · immutable
#674252 ·published 2007-08-29 01:37 UTC
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asterisk1*CLI>
<-- SIP read from 208.239.76.170:5060:
INVITE sip:s@38.113.114.224;useradd=38.113.114.224;userport=5060 SIP/2.0
Via: SIP/2.0/UDP 208.239.76.170:5060;branch=z9hG4bK0o43bu100051njgoc2g0.1
From: "unknown" <sip:3104991266@10.0.5.25>;tag=SDokoo601-as058ae75d
To: <sip:7665699-tr5jg2eb53ne8@10.0.5.60>
Contact: <sip:3104991266@208.239.76.170:5060;transport=udp>
Call-ID: SDokoo601-6ea3b21e52497ba0008620fa47176475-gurp4l2
CSeq: 102 INVITE
User-Agent: IPX Softswitch 2.0
Max-Forwards: 69
Remote-Party-ID: "unknown" <sip:3104991266@10.0.5.25>;privacy=off;screen=no
Date: Wed, 29 Aug 2007 01:33:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-UUID: 1188351203.899855.5302
Content-Type: application/sdp
Content-Length: 556

v=0
o=root 7130 7130 IN IP4 208.239.76.170
s=session
c=IN IP4 208.239.76.170
t=0 0
m=audio 56864 RTP/AVP 3 4 18 111 97 0 8 10 101
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:111 G726-32/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:10 L16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
m=video 63518 RTP/AVP 26 103 34 31
a=rtpmap:26 JPEG/90000
a=rtpmap:103 h263-1998/90000
a=rtpmap:34 H263/90000
a=rtpmap:31 H261/90000

--- (15 headers 23 lines) ---
Using INVITE request as basis request - SDokoo601-6ea3b21e52497ba0008620fa47176475-gurp4l2
Sending to 208.239.76.170 : 5060 (non-NAT)
Found peer '8182323408_OUT'
Found RTP audio format 3
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 111
Found RTP audio format 97
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 10
Found RTP audio format 101
Peer audio RTP is at port 208.239.76.170:56864
Found description format GSM
Found description format G723
Found description format G729
Found description format G726-32
Found description format iLBC
Found description format PCMU
Found description format PCMA
Found description format L16
Found description format telephone-event
Found description format JPEG
Found description format h263-1998
Found description format H263
Found description format H261
Capabilities: us - 0x107 (g723|gsm|ulaw|g729), peer - audio=0x55f (g723|gsm|ulaw|alaw|g726|slin|g729|ilbc)/video=0x0 (nothing), combined - 0x107 (g723|gsm|ulaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for s in from-pstn (domain 38.113.114.224;useradd=38.113.114.224;userport=5060)
list_route: hop: <sip:3104991266@208.239.76.170:5060;transport=udp>
Transmitting (no NAT) to 208.239.76.170:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 208.239.76.170:5060;branch=z9hG4bK0o43bu100051njgoc2g0.1;received=208.239.76.170
From: "unknown" <sip:3104991266@10.0.5.25>;tag=SDokoo601-as058ae75d
To: <sip:7665699-tr5jg2eb53ne8@10.0.5.60>
Call-ID: SDokoo601-6ea3b21e52497ba0008620fa47176475-gurp4l2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@38.113.114.224>
Content-Length: 0


---
    -- Executing NoOp("SIP/7665699-b79043c0", "No DID or CID Match") in new stack
    -- Executing Answer("SIP/7665699-b79043c0", "") in new stack
We're at 38.113.114.224 port 12220
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 208.239.76.170:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.239.76.170:5060;branch=z9hG4bK0o43bu100051njgoc2g0.1;received=208.239.76.170
From: "unknown" <sip:3104991266@10.0.5.25>;tag=SDokoo601-as058ae75d
To: <sip:7665699-tr5jg2eb53ne8@10.0.5.60>;tag=as10af7068
Call-ID: SDokoo601-6ea3b21e52497ba0008620fa47176475-gurp4l2
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@38.113.114.224>
Content-Type: application/sdp
Content-Length: 312

v=0
o=root 6417 6417 IN IP4 38.113.114.224
s=session
c=IN IP4 38.113.114.224
t=0 0
m=audio 12220 RTP/AVP 0 4 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Executing Wait("SIP/7665699-b79043c0", "2") in new stack
asterisk1*CLI>
<-- SIP read from 208.239.76.170:5060:
ACK sip:s@38.113.114.224 SIP/2.0
Via: SIP/2.0/UDP 208.239.76.170:5060;branch=z9hG4bK0o43bu1010shhd0bv7g0.1
From: "unknown" <sip:3104991266@10.0.5.25>;tag=SDokoo601-as058ae75d
To: <sip:7665699-tr5jg2eb53ne8@10.0.5.60>;tag=as10af7068
Contact: <sip:3104991266@208.239.76.170:5060;transport=udp>
Call-ID: SDokoo601-6ea3b21e52497ba0008620fa47176475-gurp4l2
CSeq: 102 ACK
User-Agent: IPX Softswitch 2.0
Max-Forwards: 69
Remote-Party-ID: "unknown" <sip:3104991266@10.0.5.25>;privacy=off;screen=no
Content-Length: 0


--- (11 headers 0 lines) ---
    -- Executing Playback("SIP/7665699-b79043c0", "ss-noservice") in new stack
    -- Playing 'ss-noservice' (language 'en')
    -- Executing SayAlpha("SIP/7665699-b79043c0", "") in new stack
12 headers, 0 lines
Reliably Transmitting (NAT) to 76.175.234.137:5060:
OPTIONS sip:9000@192.168.1.137:5060;rinstance=1fcecec2909e2ce2 SIP/2.0
Via: SIP/2.0/UDP 38.113.114.224:5060;branch=z9hG4bK6edc8add;rport
From: "Unknown" <sip:Unknown@38.113.114.224>;tag=as5759b581
To: <sip:9000@192.168.1.137:5060;rinstance=1fcecec2909e2ce2>
Contact: <sip:Unknown@38.113.114.224>
Call-ID: 332d6c18319b90b04320551732c5fab7@38.113.114.224
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 29 Aug 2007 01:33:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
asterisk1*CLI>
<-- SIP read from 76.175.234.137:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 38.113.114.224:5060;branch=z9hG4bK6edc8add;rport=5060
Contact: <sip:192.168.1.137:5060>
To: <sip:9000@192.168.1.137:5060;rinstance=1fcecec2909e2ce2>;tag=b270d37b
From: "Unknown"<sip:Unknown@38.113.114.224>;tag=as5759b581
Call-ID: 332d6c18319b90b04320551732c5fab7@38.113.114.224
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS
User-Agent: Idefisk
Allow-Events: message-summary
Content-Length: 0


--- (13 headers 0 lines) ---
Destroying call '332d6c18319b90b04320551732c5fab7@38.113.114.224'
Scheduling destruction of call 'SDokoo601-6ea3b21e52497ba0008620fa47176475-gurp4l2' in 32000 ms
set_destination: Parsing <sip:3104991266@208.239.76.170:5060;transport=udp> for address/port to send to
set_destination: set destination to 208.239.76.170, port 5060
Reliably Transmitting (no NAT) to 208.239.76.170:5060:
BYE sip:3104991266@208.239.76.170:5060 SIP/2.0
Via: SIP/2.0/UDP 38.113.114.224:5060;branch=z9hG4bK7148ffb2;rport
From: <sip:7665699-tr5jg2eb53ne8@10.0.5.60>;tag=as10af7068
To: "unknown" <sip:3104991266@10.0.5.25>;tag=SDokoo601-as058ae75d
Call-ID: SDokoo601-6ea3b21e52497ba0008620fa47176475-gurp4l2
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
asterisk1*CLI>
<-- SIP read from 208.239.76.170:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 38.113.114.224:5060;received=38.113.114.224;branch=z9hG4bK7148ffb2;rport=5060
From: <sip:7665699-tr5jg2eb53ne8@10.0.5.60>;tag=as10af7068
To: "unknown" <sip:3104991266@10.0.5.25>;tag=SDokoo601-as058ae75d
Call-ID: SDokoo601-6ea3b21e52497ba0008620fa47176475-gurp4l2
CSeq: 102 BYE
User-Agent: IPX Softswitch 2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:3104991266@208.239.76.170:5060;transport=udp>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


--- (11 headers 0 lines) ---
Destroying call 'SDokoo601-6ea3b21e52497ba0008620fa47176475-gurp4l2'
asterisk1*CLI>
<-- SIP read from 76.175.234.137:5060:



--- (0 headers 1 lines) ---