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=as630fa638;lr=on>
Via: SIP/2.0/UDP 4.79.19.56:5060;branch=z9hG4bKdf8e.ee6f43d7.0
Via: SIP/2.0/UDP 4.79.19.58:5060;received=192.168.3.5;branch=z9hG4bK62137a27;rport=5060
From: "6504300079" <sip:6504300079@4.79.19.58>;tag=as630fa638
To: <sip:4086241467@proxy-in.telasip.com;user=phone>
Contact: <sip:6504300079@192.168.3.5:5060>
Call-ID: 126bfd482c0d8b166806ff4545b1fcd5@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 19:20:31 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 15062 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 - 126bfd482c0d8b166806ff4545b1fcd5@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 4.79.19.58:15062
Found description format PCMU
Found description format G726-32
Found description format G729
Found description format GSM
Found description format telephone-event
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x116 (gsm|ulaw|g726|g729)/video=0x0 (not hing), combined - 0x6 (gsm|ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combine d - 0x1 (telephone-event)
Looking for 4086241467 in from-trunk (domain 38.99.92.35)
list_route: hop: <sip:4.79.19.56;ftag=as630fa638;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=z9hG4bKdf8e.ee6f43d7.0;received=4.79.19.56
Via: SIP/2.0/UDP 4.79.19.58:5060;received=192.168.3.5;branch=z9hG4bK62137a27;rport=5060
From: "6504300079" <sip:6504300079@4.79.19.58>;tag=as630fa638
To: <sip:4086241467@proxy-in.telasip.com;user=phone>
Call-ID: 126bfd482c0d8b166806ff4545b1fcd5@4.79.19.58
Seq: 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
---
We're at 38.99.92.35 port 14974
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) 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=z9hG4bKdf8e.ee6f43d7.0;received=4.79.19.56
Via: SIP/2.0/UDP 4.79.19.58:5060;received=192.168.3.5;branch=z9hG4bK62137a27;rport=5060
Record-Route: <sip:4.79.19.56;ftag=as630fa638;lr=on>
From: "6504300079" <sip:6504300079@4.79.19.58>;tag=as630fa638
To: <sip:4086241467@proxy-in.telasip.com;user=phone>;tag=as140388e2
Call-ID: 126bfd482c0d8b166806ff4545b1fcd5@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: 235
v=0
o=root 4463 4463 IN IP4 38.99.92.35
s=session
c=IN IP4 38.99.92.35
t=0 0
m=audio 14974 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 - - - -
---
== Spawn extension (from-trunk, 4086241467, 4) exited non-zero on 'SIP/mgraham2-096142a 0'
Scheduling destruction of call '126bfd482c0d8b166806ff4545b1fcd5@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=as630fa638;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=z9hG4bK032b6d68;rport=5060
From: "6504300079" <sip:6504300079@4.79.19.58>;tag=as630fa638
To: <sip:4086241467@proxy-in.telasip.com;user=phone>;tag=as140388e2
Contact: <sip:6504300079@4.79.19.58>
Call-ID: 126bfd482c0d8b166806ff4545b1fcd5@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=as630fa638;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=z9hG4bK6efea1a4;rport
Route: <sip:4.79.19.56;ftag=as630fa638;lr=on>
From: <sip:4086241467@proxy-in.telasip.com;user=phone>
To: "6504300079" <sip:6504300079@4.79.19.58>;tag=as630fa638
Contact: <sip:4086241467@38.99.92.35>
Call-ID: 126bfd482c0d8b166806ff4545b1fcd5@4.79.19.58
CSeq: 101 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Scheduling destruction of call '126bfd482c0d8b166806ff4545b1fcd5@4.79.19.58' in 32000 ms
<-- 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=as630fa638;lr=on>
Via: SIP/2.0/UDP 4.79.19.56:5060;branch=z9hG4bKef8e.883a9151.0
Via: SIP/2.0/UDP 4.79.19.58:5060;received=192.168.3.5;branch=z9hG4bK2df492bb;rport=5060
From: "6504300079" <sip:6504300079@4.79.19.58>;tag=as630fa638
To: <sip:4086241467@proxy-in.telasip.com;user=phone>;tag=as140388e2
Contact: <sip:6504300079@4.79.19.58>
Call-ID: 126bfd482c0d8b166806ff4545b1fcd5@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 18664 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (17 headers 10 lines) ---
Using INVITE request as basis request - 126bfd482c0d8b166806ff4545b1fcd5@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=z9hG4bKef8e.883a9151.0;received=4.79.19.56
Via: SIP/2.0/UDP 4.79.19.58:5060;received=192.168.3.5;branch=z9hG4bK2df492bb;rport=5060
From: "6504300079" <sip:6504300079@4.79.19.58>;tag=as630fa638
To: <sip:4086241467@proxy-in.telasip.com;user=phone>;tag=as140388e2
Call-ID: 126bfd482c0d8b166806ff4545b1fcd5@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:
INVITE sip:4086241467@38.99.92.35:4068 SIP/2.0
Record-Route: <sip:4.79.19.56;ftag=as630fa638;lr=on>
Via: SIP/2.0/UDP 4.79.19.56:5060;branch=z9hG4bKef8e.883a9151.0
Via: SIP/2.0/UDP 4.79.19.58:5060;received=192.168.3.5;branch=z9hG4bK2df492bb;rport=5060
From: "6504300079" <sip:6504300079@4.79.19.58>;tag=as630fa638
To: <sip:4086241467@proxy-in.telasip.com;user=phone>;tag=as140388e2
Contact: <sip:6504300079@4.79.19.58>
Call-ID: 126bfd482c0d8b166806ff4545b1fcd5@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 18664 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) ---
Ignoring this INVITE request
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=z9hG4bKef8e.883a9151.0
From: "6504300079" <sip:6504300079@4.79.19.58>;tag=as630fa638
Call-ID: 126bfd482c0d8b166806ff4545b1fcd5@4.79.19.58
To: <sip:4086241467@proxy-in.telasip.com;user=phone>;tag=as140388e2
CSeq: 103 ACK
Route: <sip:4.79.19.56;ftag=as630fa638;lr=on>
Content-Length: 0
--- (8 headers 0 lines) ---
asterisk*CLI>
<-- 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=z9hG4bK6efea1a4;rport=4068
From: <sip:4086241467@proxy-in.telasip.com;user=phone>
To: "6504300079" <sip:6504300079@4.79.19.58>;tag=as630fa638
Call-ID: 126bfd482c0d8b166806ff4545b1fcd5@4.79.19.58
CSeq: 101 CANCEL
Content-Length: 0
Warning: 392 192.168.3.33:5060 "Noisy feedback tells: pid=18604 req_src_ip=38.99.92.35 r eq_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) ---
Destroying call '126bfd482c0d8b166806ff4545b1fcd5@4.79.19.58'
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=as630fa638;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=z9hG4bK2df492bb;rport=5060
From: "6504300079" <sip:6504300079@4.79.19.58>;tag=as630fa638
To: <sip:4086241467@proxy-in.telasip.com;user=phone>;tag=as140388e2
Contact: <sip:6504300079@4.79.19.58>
Call-ID: 126bfd482c0d8b166806ff4545b1fcd5@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=z9hG4bK2df492bb;rport=5060
From: "6504300079" <sip:6504300079@4.79.19.58>;tag=as630fa638
To: <sip:4086241467@proxy-in.telasip.com;user=phone>;tag=as140388e2
Call-ID: 126bfd482c0d8b166806ff4545b1fcd5@4.79.19.58
CSeq: 103 ACK
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Destroying call '06df827e1688ebd7075ee8a729b44c46@127.0.0.1'
asterisk*CLI>
<-- SIP read from 10.223.0.116:47968:
--- (0 headers 1 lines) ---
12 headers, 0 lines
Reliably Transmitting (no NAT) to 4.79.19.56:5060:
OPTIONS sip:gw3.sip.telasip.com SIP/2.0
Via: SIP/2.0/UDP 38.99.92.35:5060;branch=z9hG4bK1c755e4d;rport
From: "Unknown" <sip:Unknown@38.99.92.35>;tag=as6dc3e959
To: <sip:gw3.sip.telasip.com>
Contact: <sip:Unknown@38.99.92.35>
Call-ID: 4c5f7339420143863f922ef11dd4b9a7@38.99.92.35
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 15 Dec 2006 19:20:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk*CLI>
<-- SIP read from 4.79.19.56:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 38.99.92.35:5060;branch=z9hG4bK1c755e4d;rport=4068
From: "Unknown" <sip:Unknown@38.99.92.35>;tag=as6dc3e959
To: <sip:gw3.sip.telasip.com>;tag=9baf1a1f504a9f775e8d7e67a181fc72.bfb9
Call-ID: 4c5f7339420143863f922ef11dd4b9a7@38.99.92.35
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
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=sip:gw3.sip.telasip.com out_uri=sip:gw3.sip.telasip.com via_cnt==1"
--- (12 headers 0 lines) ---
Destroying call '4c5f7339420143863f922ef11dd4b9a7@38.99.92.35'
asterisk*CLI>