[Asterisk-Users] Sip phone will not connect

Tommy Denton tommydenton at gmail.com
Mon Sep 12 00:50:53 MST 2005


Folks,

I have made an inbound call. Now I want to make an out bound.

I have a sip softphone hitting the pbx. On my motorola firewall I have 
allowed all traffic in and out to my address. There is not a firewall on the 
PBX yet.

here is the log off the softphone..and the PBX logs look like this 

Registration from 'Tommy Denton <sip:201 at 209.101.93.30>' failed for '
24.0.114.xxx'



(c) 2004 Xten Networks, Inc. All rights reserved.
X-Lite release 1103m build stamp 14262
License key: 258C984DF72244D39564431814E958A1

Established SIP protocol listen on: 192.168.10.8:5060<http://192.168.10.8:5060>

Discovered Port Restricted Cone NAT Firewall

SIP: 192.168.10.8:5060 <http://192.168.10.8:5060>
RTP: 192.168.10.8:8000 <http://192.168.10.8:8000>
NAT: 24.0.114.xxx

PROXY#0: 209.101.93.30:5060 <http://209.101.93.30:5060>


SEND TIME: 1583837
SEND >> 209.101.93.30:5060 <http://209.101.93.30:5060>
REGISTER sip:209.101.93.30 <http://209.101.93.30> SIP/2.0
Via: SIP/2.0/UDP 24.0.114.xxx
:5060;rport;branch=z9hG4bK3FCFB2074CD548A3A36203C12103C909
From: Tommy Denton <sip:201 at 209.101.93.30>;tag=994123068
To: Tommy Denton <sip:201 at 209.101.93.30>
Contact: "Tommy Denton" <sip:201 at 24.0.114.xxx:5060>
Call-ID: 017C3F45DBEA4175A2FAC7D2DB6EB08C at 209.101.93.30
CSeq: 60444 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0


RECEIVE TIME: 1583887
RECEIVE << 209.101.93.30:5060 <http://209.101.93.30:5060>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 24.0.114.xxx
:5060;branch=z9hG4bK3FCFB2074CD548A3A36203C12103C909
From: Tommy Denton <sip:201 at 209.101.93.30>;tag=994123068
To: Tommy Denton <sip:201 at 209.101.93.30>
Call-ID: 017C3F45DBEA4175A2FAC7D2DB6EB08C at 209.101.93.30
CSeq: 60444 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:201 at 209.101.93.30>
Content-Length: 0


RECEIVE TIME: 1583887
RECEIVE << 209.101.93.30:5060 <http://209.101.93.30:5060>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 24.0.114.xxx
:5060;branch=z9hG4bK3FCFB2074CD548A3A36203C12103C909
From: Tommy Denton <sip:201 at 209.101.93.30>;tag=994123068
To: Tommy Denton <sip:201 at 209.101.93.30>;tag=as3cf55c89
Call-ID: 017C3F45DBEA4175A2FAC7D2DB6EB08C at 209.101.93.30
CSeq: 60444 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:201 at 209.101.93.30>
WWW-Authenticate: Digest realm="asterisk", nonce="501ab07e"
Content-Length: 0


SEND TIME: 1583897
SEND >> 209.101.93.30:5060 <http://209.101.93.30:5060>
REGISTER sip:209.101.93.30 <http://209.101.93.30> SIP/2.0
Via: SIP/2.0/UDP 24.0.114.xxx
:5060;rport;branch=z9hG4bK54AD707DDFB94A98A48DFC69DB4264B0
From: Tommy Denton <sip:201 at 209.101.93.30>;tag=994123068
To: Tommy Denton <sip:201 at 209.101.93.30>
Contact: "Tommy Denton" <sip:201 at 24.0.114.xxx:5060>
Call-ID: 017C3F45DBEA4175A2FAC7D2DB6EB08C at 209.101.93.30
CSeq: 60445 REGISTER
Expires: 1800
Authorization: Digest 
username="201",realm="asterisk",nonce="501ab07e",response="7485b19096c388ced23d5ba6c6d47c3b",uri="sip:
209.101.93.30 <http://209.101.93.30>"
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0
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