[jr]

Ik wil Asterisk configureren als SIP client, waarbij binnenkomende gesprekken op mijn sipgate account worden doorgesluisd naar een locale user (6000). Ik heb sipgate.de geconfigureerd en volgens mij gaat de registratie goed:

Host dnsmgr Username Refresh State
Reg.Time
sipgate.de:5060 N 2394288 1785 Registered
Sun, 27 Sep 2009 16:21:34
1 SIP registrations.

Als ik de peers opvraag, zie ik ook user 6000:
Name/username Host Dyn Nat ACL Port Status
2394288/2394288 217.10.79.9 N 5060 Unmonitored
6000/xlite 10.164.4.46 D 50886 Unmonitored
incoming 217.10.79.9 5060 Unmonitored
3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0 offline]

Ik heb SIP messages met debug info opgevraagd, en zie het volgende verschijnen als ik naar mijn sipgate nummer bel:

<--- SIP read from UDP://217.10.79.9:5060 --->
INVITE sip:2394288@10.131.1.0:5060 SIP/2.0
Record-Route: <sip:217.10.79.9;lr=on;ftag=as2fe2de1e>
Record-Route: <sip:172.20.40.2;lr=on>
Record-Route: <sip:217.10.79.9;lr=on;ftag=as2fe2de1e>
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK8659.b7356ef2.0
Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK8659.b7356ef2.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.226;branch=z9hG4bK6115a311
Via: SIP/2.0/UDP 217.10.67.13:5060;branch=z9hG4bK6115a311;rport=5060
From: "<mobile phone nr>" <sip:<mobile phone nr>@sipgate.de>;tag=as2fe2de1e
To: <sip:004924153807890@sipgate.de>
Contact: <sip:<mobile phone nr>@217.10.67.13>
Call-ID: 6f21e9585b435f3c0a22ea8b0a7445ab@sipgate.de
CSeq: 102 INVITE
Max-Forwards: 67
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 363

v=0
o=root 22417 22417 IN IP4 217.10.67.13
s=session
c=IN IP4 217.10.67.13
t=0 0
m=audio 15022 RTP/AVP 8 0 3 97 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
--- (18 headers 17 lines) ---
== Using SIP RTP CoS mark 5
Sending to 217.10.79.9 : 5060 (no NAT)
Using INVITE request as basis request - 6f21e9585b435f3c0a22ea8b0a7445ab@sipgate
.de
Found peer '2394288' for '<mobile phone nr>' from 217.10.79.9:5060
ac-vm-brink-linux1*CLI>
<--- Reliably Transmitting (NAT) to 217.10.79.9:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK8659.b7356ef2.0;received=217.10.
79.9
Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK8659.b7356ef2.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.226;branch=z9hG4bK6115a311
Via: SIP/2.0/UDP 217.10.67.13:5060;branch=z9hG4bK6115a311;rport=5060
From: "<mobile phone nr>" <sip:<mobile phone nr>@sipgate.de>;tag=as2fe2de1e
To: <sip:004924153807890@sipgate.de>;tag=as0abe516b
Call-ID: 6f21e9585b435f3c0a22ea8b0a7445ab@sipgate.de
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.1.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="34ff43e4"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '6f21e9585b435f3c0a22ea8b0a7445ab@sipgate.d
e' in 32000 ms (Method: INVITE)
ac-vm-brink-linux1*CLI>
<--- SIP read from UDP://217.10.79.9:5060 --->
ACK sip:2394288@10.131.1.0:5060 SIP/2.0
Max-Forwards: 10
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK8659.b7356ef2.0
Via: SIP/2.0/UDP 172.20.40.2;branch=z9hG4bK8659.b7356ef2.0
From: "<mobile phone nr>" <sip:<mobile phone nr>@sipgate.de>;tag=as2fe2de1e
Call-ID: 6f21e9585b435f3c0a22ea8b0a7445ab@sipgate.de
To: <sip:004924153807890@sipgate.de>;tag=as0abe516b
CSeq: 102 ACK
Content-Length: 0
X-hint: rr-enforced


<------------->
--- (10 headers 0 lines) ---

Mijn extensions.conf:
[code]
[general]
exten => _0.,1,NoOp(Call via sipgate.de)
exten => _0.,2, Ringing
exten => _0.,3,SetCallerID(Jan van den Brink)
exten => _0.,4,Dial(SIP/${EXTEN:1}@2394288,60)
exten => _0.,5,Hangup

[incoming]
exten => 2394288,1,NoOp(--- ${CALLERID} calling on sipgate (${EXTEN}) ---)
exten => 2394288,2,Ringing
exten => 2394288,3,Wait(1)
exten => 2394288,4,Dial(SIP/6000,30)

Mijn sip.conf:

[code]
[general]
port=5060
bindaddr=0.0.0.0
context=default
register=>2394288:<password>@sipgate.de/2394288

[2394288]
type=peer
username=2394288
fromuser=2394288
secret=<password>
context=default
host=sipgate.de
fromdomain=sipgate.de
insecure=port,invite
caninvite=no
canreinvite=no
nat=yes
externip=62.134.199.226
localnet=10.131.1.0/255.255.254.0
quality=yes
disallow=all
allow=ulaw
allow=alaw

[incoming]
type=peer
fromdomain=sipgate.de
host=sipgate.de
context=incoming
disallow=all
allow=ulaw
allow=alaw
[/code]

Ik zie dus dat mijn belletje binnenkomt in Asterisk, maar om de een of andere reden zie ik ook een 401 error. Verder zie ik niet dat user 6000 gebeld wordt (hij komt niet in context voor incoming??). Elke hulp/tip is van harte welkom!!