All pastes #280313 Raw Edit

Mine

public text v1 · immutable
·published 2006-12-15 20:35 UTC
rendered paste body
<-- SIP read from 4.79.19.56:5060:
INVITE sip:4086241467@38.99.92.35 SIP/2.0
Record-Route: <sip:4.79.19.56;ftag=as01d18b81;lr=on>
Via: SIP/2.0/UDP 4.79.19.56:5060;branch=z9hG4bK0656.ab98f7c4.0
Via: SIP/2.0/UDP 4.79.19.58:5060;received=192.168.3.5;branch=z9hG4bK418cef45;rport=5060
From: "6504300079" <sip:6504300079@4.79.19.58>;tag=as01d18b81
To: <sip:4086241467@proxy-in.telasip.com;user=phone>
Contact: <sip:6504300079@192.168.3.5:5060>
Call-ID: 5d27126c7bc949860bdfa4c13db8f1da@4.79.19.58
CSeq: 102 INVITE
User-Agent: Telasip GW3
Max-Forwards: 69
Remote-Party-ID: "6504300079" <sip:6504300079@4.79.19.58>;privacy=off;screen=no
Date: Fri, 15 Dec 2006 20:35:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 311
P-hint: proxy userloc applied
P-hint: e164 requested
P-hint: NAT Traversal Fixed Contact Force rport
P-hint: NAT Traversal Fixed Nated Contact

v=0
o=root 9156 9156 IN IP4 4.79.19.58
s=session
c=IN IP4 4.79.19.58
t=0 0
m=audio 11474 RTP/AVP 0 111 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
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 - - - -

--- (20 headers 14 lines) ---
Using INVITE request as basis request - 5d27126c7bc949860bdfa4c13db8f1da@4.79.19.58
Sending to 4.79.19.56 : 5060 (non-NAT)
Found peer 'telasip-gw'
Found RTP audio format 0
Found RTP audio format 111
Found RTP audio format 18
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 4.79.19.58:11474
Found description format PCMU
Found description format G726-32
Found description format G729
Found description format GSM
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x116 (gsm|ulaw|g726|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 4086241467 in from-trunk (domain 38.99.92.35)
list_route: hop: <sip:4.79.19.56;ftag=as01d18b81;lr=on>
Transmitting (no NAT) to 4.79.19.56:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 4.79.19.56:5060;branch=z9hG4bK0656.ab98f7c4.0;received=4.79.19.56
Via: SIP/2.0/UDP 4.79.19.58:5060;received=192.168.3.5;branch=z9hG4bK418cef45;rport=5060
From: "6504300079" <sip:6504300079@4.79.19.58>;tag=as01d18b81
To: <sip:4086241467@proxy-in.telasip.com;user=phone>
Call-ID: 5d27126c7bc949860bdfa4c13db8f1da@4.79.19.58
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:4086241467@38.99.92.35>
Content-Length: 0


---
    -- Executing Set("SIP/mgraham2-097239e0", "FROM_DID=4086241467") in new stack
    -- Executing Answer("SIP/mgraham2-097239e0", "") in new stack
We're at 38.99.92.35 port 11726
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 4.79.19.56:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 4.79.19.56:5060;branch=z9hG4bK0656.ab98f7c4.0;received=4.79.19.56
Via: SIP/2.0/UDP 4.79.19.58:5060;received=192.168.3.5;branch=z9hG4bK418cef45;rport=5060
Record-Route: <sip:4.79.19.56;ftag=as01d18b81;lr=on>
From: "6504300079" <sip:6504300079@4.79.19.58>;tag=as01d18b81
To: <sip:4086241467@proxy-in.telasip.com;user=phone>;tag=as24d162d3
Call-ID: 5d27126c7bc949860bdfa4c13db8f1da@4.79.19.58
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:4086241467@38.99.92.35>
Content-Type: application/sdp
Content-Length: 212

v=0
o=root 5710 5710 IN IP4 38.99.92.35
s=session
c=IN IP4 38.99.92.35
t=0 0
m=audio 11726 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Executing PlayTones("SIP/mgraham2-097239e0", "ring") in new stack
  == Spawn extension (from-trunk, 4086241467, 4) exited non-zero on 'SIP/mgraham2-097239e0'
Scheduling destruction of call '5d27126c7bc949860bdfa4c13db8f1da@4.79.19.58' in 32000 ms
asterisk*CLI>
<-- SIP read from 4.79.19.56:5060:
ACK sip:4086241467@38.99.92.35:4068 SIP/2.0
Record-Route: <sip:4.79.19.56;ftag=as01d18b81;lr=on>
Via: SIP/2.0/UDP 4.79.19.56:5060;branch=0
Via: SIP/2.0/UDP 4.79.19.58:5060;received=192.168.3.5;branch=z9hG4bK24213027;rport=5060
From: "6504300079" <sip:6504300079@4.79.19.58>;tag=as01d18b81
To: <sip:4086241467@proxy-in.telasip.com;user=phone>;tag=as24d162d3
Contact: <sip:6504300079@4.79.19.58>
Call-ID: 5d27126c7bc949860bdfa4c13db8f1da@4.79.19.58
CSeq: 102 ACK
User-Agent: Telasip GW3
Max-Forwards: 69
Remote-Party-ID: "6504300079" <sip:6504300079@4.79.19.58>;privacy=off;screen=no
Content-Length: 0
P-hint: proxy loose route
P-hint: proxy loose route - forward


--- (15 headers 0 lines) ---
set_destination: Parsing <sip:4.79.19.56;ftag=as01d18b81;lr=on> for address/port to send to
set_destination: set destination to 4.79.19.56, port 5060
Reliably Transmitting (no NAT) to 4.79.19.56:5060:
CANCEL :4086241467@38.99.92.35:4068 SIP/2.0
Via: SIP/2.0/UDP 38.99.92.35:5060;branch=z9hG4bK18decc7d;rport
Route: <sip:4.79.19.56;ftag=as01d18b81;lr=on>
From: <sip:4086241467@proxy-in.telasip.com;user=phone>
To: "6504300079" <sip:6504300079@4.79.19.58>;tag=as01d18b81
Contact: <sip:4086241467@38.99.92.35>
Call-ID: 5d27126c7bc949860bdfa4c13db8f1da@4.79.19.58
CSeq: 101 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Scheduling destruction of call '5d27126c7bc949860bdfa4c13db8f1da@4.79.19.58' in 32000 ms
asterisk*CLI>
<-- SIP read from 4.79.19.56:5060:
INVITE sip:4086241467@38.99.92.35:4068 SIP/2.0
Record-Route: <sip:4.79.19.56;ftag=as01d18b81;lr=on>
Via: SIP/2.0/UDP 4.79.19.56:5060;branch=z9hG4bK1656.a5ff2f74.0
Via: SIP/2.0/UDP 4.79.19.58:5060;received=192.168.3.5;branch=z9hG4bK2e52d469;rport=5060
From: "6504300079" <sip:6504300079@4.79.19.58>;tag=as01d18b81
To: <sip:4086241467@proxy-in.telasip.com;user=phone>;tag=as24d162d3
Contact: <sip:6504300079@4.79.19.58>
Call-ID: 5d27126c7bc949860bdfa4c13db8f1da@4.79.19.58
CSeq: 103 INVITE
User-Agent: Telasip GW3
Max-Forwards: 69
Remote-Party-ID: "6504300079" <sip:6504300079@4.79.19.58>;privacy=off;screen=no
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 216
P-hint: proxy loose route
P-hint: proxy loose route - has_totag

v=0
o=root 9156 9157 IN IP4 71.16.179.175
s=session
c=IN IP4 71.16.179.175
t=0 0
m=audio 19542 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
=fmtp:101 0-16
a=silenceSupp:off - - - -

--- (17 headers 10 lines) ---
Using INVITE request as basis request - 5d27126c7bc949860bdfa4c13db8f1da@4.79.19.58
Sending to 4.79.19.56 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 4.79.19.56:5060:
SIP/2.0 503 Unavailable
Via: SIP/2.0/UDP 4.79.19.56:5060;branch=z9hG4bK1656.a5ff2f74.0;received=4.79.19.56
Via: SIP/2.0/UDP 4.79.19.58:5060;received=192.168.3.5;branch=z9hG4bK2e52d469;rport=5060
From: "6504300079" <sip:6504300079@4.79.19.58>;tag=as01d18b81
To: <sip:4086241467@proxy-in.telasip.com;user=phone>;tag=as24d162d3
Call-ID: 5d27126c7bc949860bdfa4c13db8f1da@4.79.19.58
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:4086241467@38.99.92.35>
Content-Length: 0


---
asterisk*CLI>
<-- SIP read from 4.79.19.56:5060:
ACK sip:4086241467@38.99.92.35:4068 SIP/2.0
Via: SIP/2.0/UDP 4.79.19.56:5060;branch=z9hG4bK1656.a5ff2f74.0
From: "6504300079" <sip:6504300079@4.79.19.58>;tag=as01d18b81
Call-ID: 5d27126c7bc949860bdfa4c13db8f1da@4.79.19.58
To: <sip:4086241467@proxy-in.telasip.com;user=phone>;tag=as24d162d3
CSeq: 103 ACK
Route: <sip:4.79.19.56;ftag=as01d18b81;lr=on>
Content-Length: 0


--- (8 headers 0 lines) ---

<-- SIP read from 4.79.19.56:5060:
SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL)
Via: SIP/2.0/UDP 38.99.92.35:5060;branch=z9hG4bK18decc7d;rport=4068
From: <sip:4086241467@proxy-in.telasip.com;user=phone>
To: "6504300079" <sip:6504300079@4.79.19.58>;tag=as01d18b81
Call-ID: 5d27126c7bc949860bdfa4c13db8f1da@4.79.19.58
CSeq: 101 CANCEL
Content-Length: 0
Warning: 392 192.168.3.33:5060 "Noisy feedback tells:  pid=18602 req_src_ip=38.99.92.35 req_src_port=4068 in_uri=:4086241467@38.99.92.35:4068 out_uri=:4086241467@38.99.92.35:4068 via_cnt==1"


--- (8 headers 0 lines) ---
    -- Incoming call: Got SIP response 500 "I'm terribly sorry, server error occurred (1/SL)" back from 4.79.19.56
Destroying call '5d27126c7bc949860bdfa4c13db8f1da@4.79.19.58'

<-- SIP read from 4.79.19.56:5060:
ACK sip:4086241467@38.99.92.35:4068 SIP/2.0
Record-Route: <sip:4.79.19.56;ftag=as01d18b81;lr=on>
Via: SIP/2.0/UDP 4.79.19.56:5060;branch=0
Via: SIP/2.0/UDP 4.79.19.58:5060;received=192.168.3.5;branch=z9hG4bK2e52d469;rport=5060
From: "6504300079" <sip:6504300079@4.79.19.58>;tag=as01d18b81
To: <sip:4086241467@proxy-in.telasip.com;user=phone>;tag=as24d162d3
Contact: <sip:6504300079@4.79.19.58>
Call-ID: 5d27126c7bc949860bdfa4c13db8f1da@4.79.19.58
CSeq: 103 ACK
User-Agent: Telasip GW3
Max-Forwards: 69
Remote-Party-ID: "6504300079" <sip:6504300079@4.79.19.58>;privacy=off;screen=no
Content-Length: 0
P-hint: proxy loose route
P-hint: proxy loose route - forward


--- (15 headers 0 lines) ---
Transmitting (no NAT) to 4.79.19.56:5060:
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 4.79.19.56:5060;branch=0;received=4.79.19.56
Via: SIP/2.0/UDP 4.79.19.58:5060;received=192.168.3.5;branch=z9hG4bK2e52d469;rport=5060
From: "6504300079" <sip:6504300079@4.79.19.58>;tag=as01d18b81
To: <sip:4086241467@proxy-in.telasip.com;user=phone>;tag=as24d162d3
Call-ID: 5d27126c7bc949860bdfa4c13db8f1da@4.79.19.58
CSeq: 103 ACK
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
asterisk*CLI>
<-- SIP read from 10.223.0.116:47968:



--- (0 headers 1 lines) ---
asterisk*CLI>