All pastes #740032 Raw Edit

Something

public text v1 · immutable
·published 2007-10-17 17:25 UTC
rendered paste body
voicegate*CLI>
<-- SIP read from 209.17.160.130:5060:
INVITE sip:9193833430@24.211.242.232:5060 SIP/2.0
Max-Forwards: 10
Via: SIP/2.0/UDP 209.17.160.130;branch=z9hG4bK0904.1edf4903.0
Via: SIP/2.0/UDP 216.143.130.87:5060;branch=z9hG4bK608318ec;rport=5060
From: "restricted" <sip:restricted@216.143.130.87>;tag=as5b8f34d3
To: <sip:9193833430@209.17.160.130>
Contact: <sip:restricted@216.143.130.87>
Call-ID: 24459c045dfffa6c4ee6481b060dca33@216.143.130.87
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 17 Oct 2007 14:02:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 15912 15912 IN IP4 216.143.130.87
s=session
c=IN IP4 216.143.130.87
t=0 0
m=audio 33974 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (14 headers 12 lines) ---
Using INVITE request as basis request - 24459c045dfffa6c4ee6481b060dca33@216.143.130.87
Sending to 209.17.160.130 : 5060 (NAT)
Found no matching peer or user for '209.17.160.130:5060'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 216.143.130.87:33974
Found description format PCMU
Found description format G729
Found description format telephone-event
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 9193833430 in default (domain 24.211.242.232)
Reliably Transmitting (NAT) to 209.17.160.130:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 209.17.160.130;branch=z9hG4bK0904.1edf4903.0;received=209.17.160.130
Via: SIP/2.0/UDP 216.143.130.87:5060;branch=z9hG4bK608318ec;rport=5060
From: "restricted" <sip:restricted@216.143.130.87>;tag=as5b8f34d3
To: <sip:9193833430@209.17.160.130>;tag=as7d2ff4e7
Call-ID: 24459c045dfffa6c4ee6481b060dca33@216.143.130.87
CSeq: 102 INVITE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Content-Length: 0


---

<-- SIP read from 209.17.160.130:5060:
INVITE sip:s@24.211.242.232:5060 SIP/2.0
Max-Forwards: 10
Via: SIP/2.0/UDP 209.17.160.130;branch=z9hG4bK0904.1edf4903.1
Via: SIP/2.0/UDP 216.143.130.87:5060;branch=z9hG4bK608318ec;rport=5060
From: "restricted" <sip:restricted@216.143.130.87>;tag=as5b8f34d3
To: <sip:9193833430@209.17.160.130>
Contact: <sip:restricted@216.143.130.87>
Call-ID: 24459c045dfffa6c4ee6481b060dca33@216.143.130.87
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 17 Oct 2007 14:02:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 15912 15912 IN IP4 216.143.130.87
s=session
c=IN IP4 216.143.130.87
t=0 0
m=audio 33974 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (14 headers 12 lines) ---
Ignoring this INVITE request
voicegate*CLI>
<-- SIP read from 209.17.160.130:5060:
ACK sip:9193833430@24.211.242.232:5060 SIP/2.0
Via: SIP/2.0/UDP 209.17.160.130;branch=z9hG4bK0904.1edf4903.0
From: "restricted" <sip:restricted@216.143.130.87>;tag=as5b8f34d3
Call-ID: 24459c045dfffa6c4ee6481b060dca33@216.143.130.87
To: <sip:9193833430@209.17.160.130>;tag=as7d2ff4e7
CSeq: 102 ACK
User-Agent: Sip EXpress router(0.8.14 (i386/freebsd))
Content-Length: 0

--- (8 headers 0 lines) ---
Destroying call '24459c045dfffa6c4ee6481b060dca33@216.143.130.87'
voicegate*CLI>
<-- SIP read from 209.17.160.130:5060:
INVITE sip:s@24.211.242.232:5060 SIP/2.0
Max-Forwards: 10
Via: SIP/2.0/UDP 209.17.160.130;branch=z9hG4bK0904.1edf4903.1
Via: SIP/2.0/UDP 216.143.130.87:5060;branch=z9hG4bK608318ec;rport=5060
From: "restricted" <sip:restricted@216.143.130.87>;tag=as5b8f34d3
To: <sip:9193833430@209.17.160.130>
Contact: <sip:restricted@216.143.130.87>
Call-ID: 24459c045dfffa6c4ee6481b060dca33@216.143.130.87
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 17 Oct 2007 14:02:14 GMT
llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 267

v=0
o=root 15912 15912 IN IP4 216.143.130.87
s=session
c=IN IP4 216.143.130.87
t=0 0
m=audio 33974 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (14 headers 12 lines) ---
Using INVITE request as basis request - 24459c045dfffa6c4ee6481b060dca33@216.143.130.87
Sending to 209.17.160.130 : 5060 (NAT)
Found no matching peer or user for '209.17.160.130:5060'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 216.143.130.87:33974
Found description format PCMU
Found description format G729
Found description format telephone-event
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for s in default (domain 24.211.242.232)
list_route: hop: <sip:restricted@216.143.130.87>
Transmitting (NAT) to 209.17.160.130:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 209.17.160.130;branch=z9hG4bK0904.1edf4903.1;received=209.17.160.130
Via: SIP/2.0/UDP 216.143.130.87:5060;branch=z9hG4bK608318ec;rport=5060
From: "restricted" <sip:restricted@216.143.130.87>;tag=as5b8f34d3
To: <sip:9193833430@209.17.160.130>
Call-ID: 24459c045dfffa6c4ee6481b060dca33@216.143.130.87
CSeq: 102 INVITE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:s@192.168.1.76>
Content-Length: 0


---
Oct 17 06:01:12 WARNING[57835533]: channel.c:2283 set_format: Unable to find a codec translation path from g729 to slin
Oct 17 06:01:12 WARNING[57835533]: channel.c:3698 tonepair_alloc: Unable to set 'SIP/216.143.130.87-40f14598' to signed linear format (write)
Oct 17 06:01:12 ERROR[57835533]: generator.c:93 opbx_generator_activate: Generator activation failed
Scheduling destruction of call '24459c045dfffa6c4ee6481b060dca33@216.143.130.87' in 32000 ms
Reliably Transmitting (NAT) to 209.17.160.130:5060:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 209.17.160.130;branch=z9hG4bK0904.1edf4903.1;received=209.17.160.130
Via: SIP/2.0/UDP 216.143.130.87:5060;branch=z9hG4bK608318ec;rport=5060
From: "restricted" <sip:restricted@216.143.130.87>;tag=as5b8f34d3
To: <sip:9193833430@209.17.160.130>;tag=as7ddf6ec7
Call-ID: 24459c045dfffa6c4ee6481b060dca33@216.143.130.87
CSeq: 102 INVITE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:s@192.168.1.76>
Content-Length: 0


---
voicegate*CLI>
<-- SIP read from 209.17.160.130:5060:
ACK sip:s@24.211.242.232:5060 SIP/2.0
Via: SIP/2.0/UDP 209.17.160.130;branch=z9hG4bK0904.1edf4903.1
From: "restricted" <sip:restricted@216.143.130.87>;tag=as5b8f34d3
Call-ID: 24459c045dfffa6c4ee6481b060dca33@216.143.130.87
To: <sip:9193833430@209.17.160.130>;tag=as7ddf6ec7
CSeq: 102 ACK
User-Agent: Sip EXpress router(0.8.14 (i386/freebsd))
Content-Length: 0

--- (8 headers 0 lines) ---
Destroying call '24459c045dfffa6c4ee6481b060dca33@216.143.130.87'
voicegate*CLI>