rendered paste body;
; SIP Configuration for Asterisk
;
[general]
context=inbound ; Default for incoming calls
allowguest=no ; Refuse unauthenticated users
register=foo:bar@vitelity-in2:5060
[vitelity-in2]
insecure=port,invite
type=friend
;disallow=all
allow=all
canreinvite=no
context=inbound
host=inbound2.vitelity.net
;host=64.2.142.27
defaultip=64.2.142.27
secret=bar
username=foo
[vitelity-out]
type=friend
context=outbound
host=outbound1.vitelity.net
;host=64.2.142.27
defaultip=64.2.142.27
insecure=port,invite
username=foo
fromuser=foo
secret=bar
trustrpid=yes
sendrpid=yes
allow=all
canreinvite=no
[100]
type=friend
username=100
secret=secret
host=dynamic
nat=yes
mailbox=Foo Bar@MyContext
context=MyContext
Subscribecontext=MyContext
[101]
type=friend
username=101
secret=secret
host=dynamic
nat=yes
mailbox=Foo Bar@MyContext
context=MyContext
Subscribecontext=MyContext
[102]
type=friend
username=102
secret=secret
host=dynamic
nat=yes
mailbox=Foo Bar@MyContext
context=MyContext
Subscribecontext=MyContext
[103]
type=friend
username=103
secret=secret
host=dynamic
nat=yes
mailbox=Foo Bar@MyContext
context=MyContext
Subscribecontext=MyContext
Inbound call from 8015551212:
<--- SIP read from 64.2.142.27:5060 --->
INVITE sip:3039525890@216.220.238.150 SIP/2.0
Via: SIP/2.0/UDP 64.2.142.27:5060;branch=z9hG4bK278fbad0;rport
From: "8015551212" <sip:8015551212@64.2.142.27>;tag=as2662575a
To: <sip:3039525890@216.220.238.150>
Contact: <sip:8015551212@64.2.142.27>
Call-ID: 7f3d3a0120100f794b239bf00e39dfac@64.2.142.27
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 14 Feb 2007 04:05:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 308
v=0
o=root 28155 28155 IN IP4 64.2.142.27
s=session
c=IN IP4 64.2.142.27
t=0 0
m=audio 12284 RTP/AVP 0 8 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/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 - - - -
<------------->
--- (13 headers 14 lines) ---
Sending to 64.2.142.27 : 5060 (NAT)
Using INVITE request as basis request - 7f3d3a0120100f794b239bf00e39dfac@64.2.142.27
No user '8015551212' in SIP users list
No matching peer for '8015551212' from '64.2.142.27:5060'
[Feb 13 21:00:24] NOTICE[62651]: chan_sip.c:13645 handle_request_invite: Failed to authenticate user "8015551212" <s
ip:8015551212@64.2.142.27>;tag=as2662575a
<--- Reliably Transmitting (NAT) to 64.2.142.27:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 64.2.142.27:5060;branch=z9hG4bK278fbad0;received=64.2.142.27;rport=5060
From: "8015551212" <sip:8015551212@64.2.142.27>;tag=as2662575a
To: <sip:3039525890@216.220.238.150>;tag=as142fc13c
Call-ID: 7f3d3a0120100f794b239bf00e39dfac@64.2.142.27
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '7f3d3a0120100f794b239bf00e39dfac@64.2.142.27' in 32000 ms (Method: INVITE)
<--- SIP read from 64.2.142.27:5060 --->
ACK sip:3039525890@216.220.238.150 SIP/2.0
Via: SIP/2.0/UDP 64.2.142.27:5060;branch=z9hG4bK278fbad0;rport
From: "8015551212" <sip:8015551212@64.2.142.27>;tag=as2662575a
To: <sip:3039525890@216.220.238.150>;tag=as142fc13c
Contact: <sip:8015551212@64.2.142.27>
Call-ID: 7f3d3a0120100f794b239bf00e39dfac@64.2.142.27
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '7f3d3a0120100f794b239bf00e39dfac@64.2.142.27' Method: ACK
Really destroying SIP dialog '028d145f49c027f42413216e71e92b03@216.220.238.150' Method: REGISTER