[Asterisk-Users] (no subject)

Sun Jul 10 09:55:37 MST 2005

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>

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