rendered paste body--- sip.conf ---
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
svrloookup=no
register => 31711111111@budgetphone.nl:xxxxxx:31711111111@sip.budgetphone.nl/31711111111
[budgetphone]
nat=no
type=peer
host=sip.budgetphone.nl
fromuser=31711111111
username=31711111111
secret=xxxxxx
qualify=no
fromdomain=budgetphone.nl
[31711111111]
nat=no
type=user
context=incoming
insecure=very
secret=xxxxxx
qualify=no
;[31711111111]
;nat=no
;type=friend
;context=incoming
;host=budgetphone.nl
;fromuser=31711111111
;fromdomain=budgetphone.nl
;username=31711111111
;insecure=very
;insecure=invite,port
;secret=xxxxxx
;qualify=no
;port=5060
[1000]
type=friend
context=toll-access
host=dynamic
callerid="First Extension" <1000>
canreinvite=no
disallow=all
allow=gsm
allow=alaw
qualify=no
[1002]
type=friend
context=toll-access
host=dynamic
callerid="Second Extension" <1002>
canreinvite=no
disallow=all
allow=gsm
allow=alaw
qualify=no
---- extensions.conf ----
[general]
[globals]
[extensions]
exten => _XX,1,Dial(SIP/${EXTEN})
[toll-access]
include => extensions
exten => _0XXXX.,1,Dial(SIP/${EXTEN}@budgetphone)
[incoming]
include => extensions
exten => s,1,Goto(10,1)
exten => 31711111111,1,Goto(10,1)
----- CLI output ----
(type 'help' for help)
*CLI>
<--- SIP read from 81.23.228.150:5060 --->
INVITE sip:31711111111@84.107.142.180 SIP/2.0
Record-Route: <sip:81.23.228.150;lr=on;ftag=as75a478c9;fcd=yes>
Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK105.753aeb11.0
Via: SIP/2.0/UDP 80.252.84.179:5060;branch=z9hG4bK4e40352d;rport=5060
From: "06124111111" <sip:06124111111@gw02-mci.budgetphone.nl>;tag=as75a478c9
To: <sip:31711111111@budgetphone.nl>
Contact: <sip:06124111111@80.252.84.179>
Call-ID: 4f43b3c73b550baf151d5c801a9d0f35@gw02-mci.budgetphone.nl
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 69
Date: Wed, 24 Oct 2007 14:11:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 391
v=0
o=root 17847 17847 IN IP4 80.252.84.179
s=session
c=IN IP4 81.23.228.147
t=0 0
m=audio 61948 RTP/AVP 3 18 5 8 0 97 110 101
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:5 DVI4/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
<------------->
--- (15 headers 17 lines) ---
Sending to 81.23.228.150 : 5060 (no NAT)
Using INVITE request as basis request - 4f43b3c73b550baf151d5c801a9d0f35@gw02-mci.budgetphone.nl
Found peer 'budgetphone'
<--- Reliably Transmitting (no NAT) to 81.23.228.150:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK105.753aeb11.0;received=81.23.228.150
Via: SIP/2.0/UDP 80.252.84.179:5060;branch=z9hG4bK4e40352d;rport=5060
From: "06124111111" <sip:06124111111@gw02-mci.budgetphone.nl>;tag=as75a478c9
To: <sip:31711111111@budgetphone.nl>;tag=as17533793
Call-ID: 4f43b3c73b550baf151d5c801a9d0f35@gw02-mci.budgetphone.nl
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0a76ef22"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '4f43b3c73b550baf151d5c801a9d0f35@gw02-mci.budgetphone.nl' in 32000 ms (Method: INVITE)
<--- SIP read from 81.23.228.150:5060 --->
ACK sip:31711111111@84.107.142.180 SIP/2.0
Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK105.753aeb11.0
From: "06124111111" <sip:06124111111@gw02-mci.budgetphone.nl>;tag=as75a478c9
Call-ID: 4f43b3c73b550baf151d5c801a9d0f35@gw02-mci.budgetphone.nl
To: <sip:31711111111@budgetphone.nl>;tag=as17533793
CSeq: 102 ACK
User-Agent: OpenSer (1.1.0-tls (i386/linux))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from 81.23.228.150:5060 --->
OPTIONS sip:84.107.142.180:44761 SIP/2.0
Via: SIP/2.0/UDP 81.23.228.150:5060;branch=0
From: sip:ping@81.23.228.150;tag=16a9e853
To: sip:84.107.142.180:44761
Call-ID: 54b21f65-3f287f86-016701@81.23.228.150
CSeq: 1 OPTIONS
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Looking for s in default (domain 84.107.142.180)
<--- Transmitting (no NAT) to 81.23.228.150:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 81.23.228.150:5060;branch=0;received=81.23.228.150
From: sip:ping@81.23.228.150;tag=16a9e853
To: sip:84.107.142.180:44761;tag=as22263e83
Call-ID: 54b21f65-3f287f86-016701@81.23.228.150
CSeq: 1 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Accept: application/sdp
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '54b21f65-3f287f86-016701@81.23.228.150' in 32000 ms (Method: OPTIONS)
<--- SIP read from 172.19.3.247:31854 --->
<------------->
--- (0 headers 1 lines) ---