All pastes #520415 Raw Edit

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#520415 ·published 2007-05-29 22:43 UTC
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ip-68-178-167-75*CLI>
<-- SIP read from 190.54.21.68:5060:
INVITE sip:25804759@68.178.167.75 SIP/2.0
Via: SIP/2.0/UDP 190.54.21.68:5060;branch=z9hG4bK58a70004;rport
From: "28551194" <sip:28551194@190.54.21.68>;tag=as34c2d0a8
To: <sip:25804759@68.178.167.75>
Contact: <sip:28551194@190.54.21.68>
Call-ID: 10795fa35afac7375342f809302070a0@190.54.21.68
CSeq: 102 INVITE
User-Agent: CallMe LatinIP.com
Max-Forwards: 70
Date: Tue, 29 May 2007 22:39:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 239

v=0
o=root 27824 27824 IN IP4 190.54.21.68
s=session
c=IN IP4 190.54.21.68
t=0 0
m=audio 16400 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

--- (13 headers 11 lines)---
Using INVITE request as basis request - 10795fa35afac7375342f809302070a0@190.54.21.68
Sending to 190.54.21.68 : 5060 (NAT)
Found peer '25804759'
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 190.54.21.68:16400
Found description format G729
Found description format telephone-event
Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 25804759 in from-sip-external (domain 68.178.167.75)
list_route: hop: <sip:28551194@190.54.21.68>
Transmitting (no NAT) to 190.54.21.68:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 190.54.21.68:5060;branch=z9hG4bK58a70004;rport;received=190.54.21.68
From: "28551194" <sip:28551194@190.54.21.68>;tag=as34c2d0a8
To: <sip:25804759@68.178.167.75>
Call-ID: 10795fa35afac7375342f809302070a0@190.54.21.68
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:25804759@68.178.167.75>
Content-Length: 0


---
    -- Executing NoOp("SIP/25804759-7d17", "Received incoming SIP connection from unknown peer to 25804759") in new stack
    -- Executing Set("SIP/25804759-7d17", "DID=25804759") in new stack
    -- Executing Goto("SIP/25804759-7d17", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing Ringing("SIP/25804759-7d17", "") in new stack
Transmitting (no NAT) to 190.54.21.68:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 190.54.21.68:5060;branch=z9hG4bK58a70004;rport;received=190.54.21.68
From: "28551194" <sip:28551194@190.54.21.68>;tag=as34c2d0a8
To: <sip:25804759@68.178.167.75>;tag=as4b73edcd
Call-ID: 10795fa35afac7375342f809302070a0@190.54.21.68
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:25804759@68.178.167.75>
Content-Length: 0


---
    -- Executing GotoIf("SIP/25804759-7d17", "0?from-trunk|25804759|1") in new stack
    -- Executing Set("SIP/25804759-7d17", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2007-05-29 22:42:22 UTC.
    -- Executing Answer("SIP/25804759-7d17", "") in new stack
We're at 68.178.167.75 port 15646
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 190.54.21.68:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 190.54.21.68:5060;branch=z9hG4bK58a70004;rport;received=190.54.21.68
From: "28551194" <sip:28551194@190.54.21.68>;tag=as34c2d0a8
To: <sip:25804759@68.178.167.75>;tag=as4b73edcd
Call-ID: 10795fa35afac7375342f809302070a0@190.54.21.68
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:25804759@68.178.167.75>
Content-Type: application/sdp
Content-Length: 241

v=0
o=root 26036 26036 IN IP4 68.178.167.75
s=session
c=IN IP4 68.178.167.75
t=0 0
m=audio 15646 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Executing Wait("SIP/25804759-7d17", "2") in new stack
ip-68-178-167-75*CLI>
<-- SIP read from 190.54.21.68:5060:
ACK sip:25804759@68.178.167.75 SIP/2.0
Via: SIP/2.0/UDP 190.54.21.68:5060;branch=z9hG4bK76fae3b6;rport
From: "28551194" <sip:28551194@190.54.21.68>;tag=as34c2d0a8
To: <sip:25804759@68.178.167.75>;tag=as4b73edcd
Contact: <sip:28551194@190.54.21.68>
Call-ID: 10795fa35afac7375342f809302070a0@190.54.21.68
CSeq: 102 ACK
User-Agent: CallMe LatinIP.com
Max-Forwards: 70
Content-Length: 0


--- (10 headers 0 lines)---
    -- Executing Playback("SIP/25804759-7d17", "ss-noservice") in new stack
    -- Playing 'ss-noservice' (language 'en')
    -- Executing PlayTones("SIP/25804759-7d17", "congestion") in new stack
    -- Executing Congestion("SIP/25804759-7d17", "5") in new stack
We're at 68.178.167.75 port 15646
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (no NAT) to 190.54.21.68:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 190.54.21.68:5060;branch=z9hG4bK58a70004;rport;received=190.54.21.68
From: "28551194" <sip:28551194@190.54.21.68>;tag=as34c2d0a8
To: <sip:25804759@68.178.167.75>;tag=as4b73edcd
Call-ID: 10795fa35afac7375342f809302070a0@190.54.21.68
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:25804759@68.178.167.75>
Content-Type: application/sdp
Content-Length: 241

v=0
o=root 26036 26037 IN IP4 68.178.167.75
s=session
c=IN IP4 68.178.167.75
t=0 0
m=audio 15646 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
ip-68-178-167-75*CLI>