[Asterisk-Users] Incoming calls from BudgetPhone.nl

[Peter Raaijmaker]

(this time with subject....)

Hello,

I’m trying to get Asterisk to accept incoming calls from budgetphone.nl.
When I dial my budgetphone nr on a PSTN KPN line it immediately gives a busy
tone.
I tried X-lite, which worked perfect, so my modem (with nat) probably is not
the problem.
I did a sip debug and got the following output.
Because I’m new to Asterisk I can’t get the error why this is not working. 
To me it all looks fine, no warnings or what so ever

 
The settings in sip.conf and extensions.conf are identical to those of
http://www.voip-info.org/tiki-index.php?page=Talkin2ya
 
Does anyone know what I’m doing wrong????
 
Thanks,
Peter.
 
 
-------------------------------
output of sip debug
-------------------------------
 
11 headers, 0 lines
Reliably Transmitting (no NAT) to 81.23.228.150:5060:
REGISTER sip:budgetphone.nl SIP/2.0
Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK732b82fc
From: <sip:31717110342 at budgetphone.nl>;tag=as5dc83db4
To: <sip:31717110342 at budgetphone.nl>
Call-ID: 26dfb15414601a871799536a3de1f776 at 127.0.0.1
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:31717110342 at 192.168.2.3>
Event: registration
Content-Length: 0
 
 
---
server*CLI>
<-- SIP read from 81.23.228.150:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK732b82fc
From: <sip:31717110342 at budgetphone.nl>;tag=as5dc83db4
To:
<sip:31717110342 at budgetphone.nl>;tag=9b5971f23d18872ff678d4e9dae023f8.247a
Call-ID: 26dfb15414601a871799536a3de1f776 at 127.0.0.1
CSeq: 102 REGISTER
WWW-Authenticate: Digest realm="budgetphone.nl",
nonce="42d15009299d7652e8da589cee2723af4b6a96ca"
Server: Sip EXpress router (0.8.14-5 (i386/linux))
Content-Length: 0
 
 
--- (9 headers 0 lines)---
Responding to challenge, registration to domain/host name budgetphone.nl
12 headers, 0 lines
Reliably Transmitting (no NAT) to 81.23.228.150:5060:
REGISTER sip:budgetphone.nl SIP/2.0
Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK1818547e
From: <sip:31717110342 at budgetphone.nl>;tag=as7e56000d
To: <sip:31717110342 at budgetphone.nl>
Call-ID: 26dfb15414601a871799536a3de1f776 at 127.0.0.1
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Authorization: Digest username="31717110342", realm="budgetphone.nl",
algorithm=MD5, uri="sip:budgetphone.nl",
nonce="42d15009299d7652e8da589cee2723af4b6a96ca",
response="cd69279e6a2512fd48d267ceea3394da", opaque=""
Expires: 120
Contact: <sip:31717110342 at 192.168.2.3>
Event: registration
Content-Length: 0
 
 
---
server*CLI>
<-- SIP read from 81.23.228.150:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.3:5060;branch=z9hG4bK1818547e
From: <sip:31717110342 at budgetphone.nl>;tag=as7e56000d
To:
<sip:31717110342 at budgetphone.nl>;tag=9b5971f23d18872ff678d4e9dae023f8.98b0
Call-ID: 26dfb15414601a871799536a3de1f776 at 127.0.0.1
CSeq: 103 REGISTER
Contact: <sip:31717110342 at 62.131.187.108:5060>;q=0.00;expires=120
Server: Sip EXpress router (0.8.14-5 (i386/linux))
Content-Length: 0
 
 
--- (9 headers 0 lines)---
Jul 10 18:38:04 NOTICE[26004]: chan_sip.c:8266 handle_response: Outbound
Registration: Expiry for budgetphone.nl is 120 sec (Scheduling
reregistration in 105000 ms)
Destroying call '26dfb15414601a871799536a3de1f776 at 127.0.0.1'
server*CLI>
<-- SIP read from 81.23.228.150:5060:
INVITE sip:31717110342 at 62.131.187.108:5060 SIP/2.0
Max-Forwards: 10
Record-Route: <sip:31717110342 at 81.23.228.150;ftag=as47419911;lr=on>
Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK7842.b2602997.0
Via: SIP/2.0/UDP 212.203.28.2:5060;branch=z9hG4bK2de815aa
From: "0031172651375"
<sip:0031172651375 at voipgw01.budgetphone.nl>;tag=as47419911
To: <sip:31717110342 at budgetphone.nl>
Contact: <sip:0031172651375 at 212.203.28.2>
Call-ID: 3de4e14c7400163670a44c9e3f484ff6 at voipgw01.budgetphone.nl
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sun, 10 Jul 2005 16:37:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 345
 
v=0
o=root 26318 26318 IN IP4 212.203.28.2
s=session
c=IN IP4 81.23.228.139
t=0 0
m=audio 36634 RTP/AVP 3 18 5 0 97 110 101
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=rtpmap:5 DVI4/8000
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 15 lines)---
Using INVITE request as basis request -
3de4e14c7400163670a44c9e3f484ff6 at voipgw01.budgetphone.nl
Sending to 81.23.228.150 : 5060 (NAT)
Found peer '31717110342'
Reliably Transmitting (NAT) to 81.23.228.150:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
81.23.228.150;branch=z9hG4bK7842.b2602997.0;received=81.23.228.150;rport=506
0
Via: SIP/2.0/UDP 212.203.28.2:5060;branch=z9hG4bK2de815aa
From: "0031172651375"
<sip:0031172651375 at voipgw01.budgetphone.nl>;tag=as47419911
To: <sip:31717110342 at budgetphone.nl>;tag=as3f35655f
Call-ID: 3de4e14c7400163670a44c9e3f484ff6 at voipgw01.budgetphone.nl
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:31717110342 at 192.168.2.3>
Proxy-Authenticate: Digest realm="asterisk", nonce="555b996d"
Content-Length: 0
 
 
---
Scheduling destruction of call
'3de4e14c7400163670a44c9e3f484ff6 at voipgw01.budgetphone.nl' in 15000 ms
server*CLI>
<-- SIP read from 81.23.228.150:5060:
ACK sip:31717110342 at 62.131.187.108:5060 SIP/2.0
Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK7842.b2602997.0
From: "0031172651375"
<sip:0031172651375 at voipgw01.budgetphone.nl>;tag=as47419911
Call-ID: 3de4e14c7400163670a44c9e3f484ff6 at voipgw01.budgetphone.nl
To: <sip:31717110342 at budgetphone.nl>;tag=as3f35655f
CSeq: 102 ACK
User-Agent: Sip EXpress router(0.8.14-5 (i386/linux))
Content-Length: 0
 
 
--- (8 headers 0 lines)---
Destroying call '3de4e14c7400163670a44c9e3f484ff6 at voipgw01.budgetphone.nl'
server*CLI>