[Asterisk-Users] spa-1001 not getting a dial tone on my pbx

[hank smith]

hello my friend has the proxy set up his extention set up his password set up but he isn't getting a dial tone
is there a second setting we need to put the address in?
he is going to
advenced settings
line1
and in the proxy address box he is putting the info in below is the way he has it set up
Sipura SPA Configuration

Sipura Technology Inc
Info
System
SIP
Provisioning
Regional
Line 1
User 1
User Login
basic  |
advanced

Table with 4 columns and 75 rows

Line Enable:
 yes

Streaming Audio Server (SAS)
SAS Enable:
 no
SAS DLG Refresh Intvl:
30
SAS Inbound RTP Sink:

NAT Settings
NAT Mapping Enable:
 yes
NAT Keep Alive Enable:
 yes
NAT Keep Alive Msg:
$NOTIFY
NAT Keep Alive Dest:
$PROXY

Network Settings
SIP TOS/DiffServ Value:
0x68
Network Jitter Level:
 high
RTP TOS/DiffServ Value:
0xb8

SIP Settings
SIP Port:
5060
SIP 100REL Enable:
 no
EXT SIP Port:

Auth Resync-Reboot:
 yes
SIP Debug Option:
 none

Call Feature Settings
Blind Attn-Xfer Enable:
 no
MOH Server:

Xfer When Hangup Conf:
 yes

Proxy and Registration
Proxy:
67.183.118.6
Use Outbound Proxy:
 no
Outbound Proxy:

Use OB Proxy In Dialog:
 no
Register:
 yes
Make Call Without Reg:
 no
Register Expires:
60
Ans Call Without Reg:
 no
Use DNS SRV:
 no
DNS SRV Auto Prefix:
 no
Proxy Fallback Intvl:
3600

Subscriber Information
Display Name:
Herbie Allen
User ID:
202
Password:
*************
Use Auth ID:
 yes
Auth ID:
202

Mini Certificate:

SRTP Private Key:

Supplementary Service Subscription
Call Waiting Serv:
 yes
Block CID Serv:
 yes
Block ANC Serv:
 yes
Dist Ring Serv:
 yes
Cfwd All Serv:
 yes
Cfwd Busy Serv:
 yes
Cfwd No Ans Serv:
 yes
Cfwd Sel Serv:
 yes
Cfwd Last Serv:
 yes
Block Last Serv:
 yes
Accept Last Serv:
 yes
DND Serv:
 yes
CID Serv:
 yes
CWCID Serv:
 yes
Call Return Serv:
 yes
Call Back Serv:
 yes
Three Way Call Serv:
 yes
Three Way Conf Serv:
 yes
Attn Transfer Serv:
 yes
Unattn Transfer Serv:
 yes
MWI Serv:
 yes
VMWI Serv:
 yes
Speed Dial Serv:
 yes
Secure Call Serv:
 yes
Referral Serv:
 yes
Feature Dial Serv:
 yes

Audio Configuration
Preferred Codec:
 G711u
Silence Supp Enable:
 no
Use Pref Codec Only:
 no
Silence Threshold:
 medium
G729a Enable:
 yes
Echo Canc Enable:
 yes
G723 Enable:
 yes
Echo Canc Adapt Enable:
 yes
G726-16 Enable:
 yes
Echo Supp Enable:
 yes
G726-24 Enable:
 yes
FAX CED Detect Enable:
 yes
G726-32 Enable:
 yes
FAX CNG Detect Enable:
 yes
G726-40 Enable:
 yes
FAX Passthru Codec:
 G711u
DTMF Tx Method:
 Auto
FAX Codec Symmetric:
 yes
Hook Flash Tx Method:
 None
FAX Passthru Method:
 NSE
Release Unused Codec:
 yes
FAX Process NSE:
 yes

Dial Plan
Dial Plan:
(xx.|*xx.|**xx.|#xx.)
Enable IP Dialing:
 no

FXS Port Polarity Configuration
Idle Polarity:
 Forward
Caller Conn Polarity:
 Forward
Callee Conn Polarity:
 Forward

table end

Undo All Changes

Submit All Changes

User Login
basic  |
advanced

Copyright © 2003 Sipura Technology. All Rights Reserved.

--
No virus found in this incoming message.
Checked by AVG Anti-Virus.
Version: 7.0.322 / Virus Database: 266.11.15 - Release Date: 5/22/2005

email:
hanksmith4 at earthlink.net
gmail:
hanksmith5 at gmail.com
msn messenger:
hanksmith4 at earthlink.net
aim:
hanksmith5
skype:
hanksmith5
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050523/dd7173ec/attachment.htm