WG: [Asterisk-Users] Newbie *@home + Xten.

Fri May 6 09:26:02 MST 2005

Hi Larry,

what is your * logging on the console when you connect / start with
"vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv"?

The console logging is much more "readable" then all the sip logging of
the phones.

Can you dial *43 or 1234  and whatever from the console?

regards

Manny

-----Ursprüngliche Nachricht-----
Von: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] Im Auftrag von Larry
Richardson
Gesendet: Freitag, 6. Mai 2005 17:57
An: asterisk-users at lists.digium.com
Betreff: [Asterisk-Users] Newbie *@home + Xten. 

I have d/l the iso (*@home 0.9) , built the * box and followed the
directions in the * handbook and
http://www.geekgazette.com/index.php?option=com_content&task=view&id=2&I
temid=26. 
I created extension 200 and verified that * was running fine.

Loaded Xten lite, setup the proxy for local ip (10.0.0.201) per the
handbook. After turning off the Norton Firewall protection, I am able to
start Xten and it says Logged in.

I wanted to start with the easiest thing, so I just wanted to SIP to my
local server (same net, no firewall/router issues). I can’t even get *43
or 1234 to work.  

Here is the Xten log for the 1234 call:

SEND TIME: 15532804
SEND >> 10.0.0.201:5060
INVITE sip:1234 at 10.0.0.201 SIP/2.0
Via: SIP/2.0/UDP
10.0.0.250:5060;rport;branch=z9hG4bK96200694E8A848F9808AD84E829CA819
From: rdelite <sip:200 at 10.0.0.201>;tag=3097086592
To: <sip:1234 at 10.0.0.201>
Contact: <sip:200 at 10.0.0.250:5060>
Call-ID: A26B9D85-1A5C-49DD-8507-B15B041B6C35 at 10.0.0.250
CSeq: 6629 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1103m
Content-Length: 242

v=0
o=200 15532664 15532804 IN IP4 10.0.0.250
s=X-Lite
c=IN IP4 10.0.0.250
t=0 0
m=audio 8000 RTP/AVP 3 97 110 101
a=rtpmap:3 gsm/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

RECEIVE TIME: 15532855
RECEIVE << 10.0.0.201:5060
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
10.0.0.250:5060;branch=z9hG4bK96200694E8A848F9808AD84E829CA819
From: rdelite <sip:200 at 10.0.0.201>;tag=3097086592
To: <sip:1234 at 10.0.0.201>;tag=as7a1af4a2
Call-ID: A26B9D85-1A5C-49DD-8507-B15B041B6C35 at 10.0.0.250
CSeq: 6629 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1234 at 10.0.0.201>
Proxy-Authenticate: Digest realm="asterisk", nonce="68e1ffea"
Content-Length: 0

SEND TIME: 15532855
SEND >> 10.0.0.201:5060
ACK sip:1234 at 10.0.0.201 SIP/2.0
Via: SIP/2.0/UDP
10.0.0.250:5060;rport;branch=z9hG4bK96200694E8A848F9808AD84E829CA819
From: rdelite <sip:200 at 10.0.0.201>;tag=3097086592
To: <sip:1234 at 10.0.0.201>;tag=as7a1af4a2
Contact: <sip:200 at 10.0.0.250:5060>
Call-ID: A26B9D85-1A5C-49DD-8507-B15B041B6C35 at 10.0.0.250
CSeq: 6629 ACK
Max-Forwards: 70
Content-Length: 0

SEND TIME: 15532865
SEND >> 10.0.0.201:5060
INVITE sip:1234 at 10.0.0.201 SIP/2.0
Via: SIP/2.0/UDP
10.0.0.250:5060;rport;branch=z9hG4bKECBD700714F4480D8E32F24791BB75C0
From: rdelite <sip:200 at 10.0.0.201>;tag=3097086592
To: <sip:1234 at 10.0.0.201>
Contact: <sip:200 at 10.0.0.250:5060>
Call-ID: A26B9D85-1A5C-49DD-8507-B15B041B6C35 at 10.0.0.250
CSeq: 6630 INVITE
Proxy-Authorization: Digest
username="200",realm="asterisk",nonce="68e1ffea",response="e4d2c26ee42a2
32380884715ea76ac71",uri="sip:1234 at 10.0.0.201"
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1103m
Content-Length: 242

v=0
o=200 15532664 15532804 IN IP4 10.0.0.250
s=X-Lite
c=IN IP4 10.0.0.250
t=0 0
m=audio 8000 RTP/AVP 3 97 110 101
a=rtpmap:3 gsm/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

RECEIVE TIME: 15532865
RECEIVE << 10.0.0.201:5060
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP
10.0.0.250:5060;branch=z9hG4bKECBD700714F4480D8E32F24791BB75C0
From: rdelite <sip:200 at 10.0.0.201>;tag=3097086592
To: <sip:1234 at 10.0.0.201>;tag=as7a1af4a2
Call-ID: A26B9D85-1A5C-49DD-8507-B15B041B6C35 at 10.0.0.250
CSeq: 6630 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1234 at 10.0.0.201>
Content-Length: 0
Larry Richardson

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