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Thank you,<br>
<br>
>>How are you setting up xlite and the ata?<br>
<br>
Xlite<br>
<br>
User name : 1000<br>
Domain: IP of the server running Asterisk<br>
Register with domain and receive incoming calls: clear<br>
Port used in local computer : manually specify range : 5061-5062<br>
<br>
ATA<br>
SIP server address: IP of the server running Asterisk<br>
Outbond Proxy : IP of the server running Asterisk<br>
SIP User id : 1001<br>
Accoount ID : 1001<br>
Use DNS SRV : yes<br>
User id is phone number : yes<br>
SIP registration : no<br>
Local sip port : 5062<br>
<br>
>>Which version of Asterisk are you using?<br>
Asterisk 1.6.1.6, Copyright (C) 1999 - 2009 Digium,<br>
<br>
>> What does the general section of your sip.conf look like?<br>
<br>
[general]<br>
context=default<br>
allowoverlap=no<br>
bindport=5060<br>
bindaddr=0.0.0.0<br>
srvlookup=yes<br>
<br>
When I <br>
<br>
sip show peers<br>
<br>
Name/username Host Dyn Nat ACL Port
Status <br>
1000 (Unspecified) D 5060
Unmonitored <br>
1001 (Unspecified) D 5060
Unmonitored <br>
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0
offline]<br>
<br>
regards<br>
<br>
Jair Santos<br>
<br>
<br>
<br>
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<br>
Warren Selby wrote:
<blockquote
cite="mid:e7f1293f0910301156i7684042bt4efdca62576325e8@mail.gmail.com"
type="cite">How are you setting up xlite and the ata? Which version
of Asterisk are you using? What does the general section of your
sip.conf look like?<br>
<br>
<div class="gmail_quote">On Fri, Oct 30, 2009 at 1:01 PM, Cliconnect <span
dir="ltr"><<a moz-do-not-send="true"
href="mailto:cliconnect@cliconnect.com">cliconnect@cliconnect.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote"
style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div bgcolor="#ffffff" text="#000066">Hi all,<br>
<br>
I can only get a line signal when I set the phones to not register
with domain . <br>
<br>
All phones are in the same NAT and I cannot make calls.<br>
<br>
I am getting "Call failed : Proxy Authentication Required" in Xlite
and a busy signal when using an ATA.<br>
<br>
Here is my extensions.conf<br>
[internal]<br>
exten => 1000,1,Verbose(1|Extension 1000)<br>
;exten => 1000,n,Echo()<br>
;exten => 1000,n,Hangup()<br>
exten => 1000,n,Dial(SIP/1000,30)<br>
exten => 1000,n,Hangup()<br>
<br>
exten => 1001,1,Verbose(1|Extension 1001)<br>
exten => 1001,n,Dial(SIP/1001,30)<br>
exten => 1001,n,Hangup()<br>
<br>
[phones]<br>
include => internal<br>
<br>
<br>
and sip.conf<br>
[1000]<br>
type=friend<br>
context=phones<br>
host=dynamic<br>
[1001]<br>
type=friend<br>
context=phones<br>
host=dynamic<br>
<br>
<br>
I am not setting a password .<br>
<br>
Any help will be appreciated.<br>
<br>
TIA<br>
<br>
Jair Santos<br>
<div>-- <br>
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