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@budgetphone.nl>;tag=as5dc83db4
To: <sip:31717110342@budgetphone.nl>
Call-ID: 26dfb15414601a871799536a3de1f776@127.0.0.1
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:31717110342@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@budgetphone.nl>;tag=as5dc83db4
To:
<sip:31717110342@budgetphone.nl>;tag=9b5971f23d18872ff678d4e9dae023f8.247a
Call-ID: 26dfb15414601a871799536a3de1f776@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@budgetphone.nl>;tag=as7e56000d
To: <sip:31717110342@budgetphone.nl>
Call-ID: 26dfb15414601a871799536a3de1f776@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@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@budgetphone.nl>;tag=as7e56000d
To:
<sip:31717110342@budgetphone.nl>;tag=9b5971f23d18872ff678d4e9dae023f8.98b0
Call-ID: 26dfb15414601a871799536a3de1f776@127.0.0.1
CSeq: 103 REGISTER
Contact: <sip:31717110342@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@127.0.0.1'
server*CLI>
<-- SIP read from 81.23.228.150:5060:
INVITE sip:31717110342@62.131.187.108:5060 SIP/2.0
Max-Forwards: 10
Record-Route: <sip:31717110342@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@voipgw01.budgetphone.nl>;tag=as47419911
To: <sip:31717110342@budgetphone.nl>
Contact: <sip:0031172651375@212.203.28.2>
Call-ID: 3de4e14c7400163670a44c9e3f484ff6@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@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@voipgw01.budgetphone.nl>;tag=as47419911
To: <sip:31717110342@budgetphone.nl>;tag=as3f35655f
Call-ID: 3de4e14c7400163670a44c9e3f484ff6@voipgw01.budgetphone.nl
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:31717110342@192.168.2.3>
Proxy-Authenticate: Digest realm="asterisk",
nonce="555b996d"
Content-Length: 0
---
Scheduling destruction of call
'3de4e14c7400163670a44c9e3f484ff6@voipgw01.budgetphone.nl' in 15000 ms
server*CLI>
<-- SIP read from 81.23.228.150:5060:
ACK sip:31717110342@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@voipgw01.budgetphone.nl>;tag=as47419911
Call-ID: 3de4e14c7400163670a44c9e3f484ff6@voipgw01.budgetphone.nl
To: <sip:31717110342@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@voipgw01.budgetphone.nl'
server*CLI>
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20050710/4764d211/attachment.htm
