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color="#000000">Thank you ,<br>
<br>
I did that <br>
sip set debug ip 198.163.0.103<br>
SIP Debugging Enabled for IP: 198.163.0.103<br>
<br>
I checked the /var/log/asterisk files and there is no information
there. <br>
<br>
Could you please inform where am I suppose to see the debug information
?<br>
<br>
tks<br>
<br>
Jair<br>
</font></div>
<br>
<br>
Ott Rose wrote:
<blockquote cite="mid:BLU137-W74FD1E81BD36468B612F4BBB60@phx.gbl"
type="cite">
<style><!--
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{
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body.hmmessage
{
font-size: 10pt;
font-family:Verdana
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--></style>you
can get debug info a couple of ways from the asterisk CLI. I like this
command the best. sip set debug ip xxx.xxx.xx.xxx where xxx.xxx.xxx.xxx
is the of the x-lite phone. It will give you a lot of info. I haven't
figured out how to redirect output yet.<br>
<br>
<hr id="stopSpelling">Date: Fri, 30 Oct 2009 13:05:35 -0700<br>
From: <a class="moz-txt-link-abbreviated" href="mailto:cliconnect@cliconnect.com">cliconnect@cliconnect.com</a><br>
To: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>
Subject: Re: [asterisk-users] Cannot make calls<br>
<br>
<br>
No change on it.<br>
<br>
Do I have to enter a command ?<br>
<br>
I have changed the port to 5060 in both clients. Still the same problem.<br>
<br>
thanks<br>
<br>
Jair Santos<br>
<br>
Warren Selby wrote:
<blockquote
cite="mid:e7f1293f0910301253x155bb04bwe193b54ee051e401@mail.gmail.com">You're
attempting to connect on ports 5061-5062 but are
bound to port 5060...?<br>
<br>
What does your CLI look like during a failed call attempt?<br>
<br>
Thanks,<br>
--Warren Selby<br>
<br>
<div class="ecxgmail_quote">On Fri, Oct 30, 2009 at 2:18 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="ecxgmail_quote" style="padding-left: 1ex;">
<div> <br>
<div><br>
</div>
Thank you,
<div class="ecxim"><br>
<br>
>>How are you setting up xlite and the ata?<br>
<br>
</div>
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
<div class="ecxim"><br>
<br>
>>Which version of Asterisk are you using?<br>
</div>
Asterisk 1.6.1.6, Copyright (C) 1999 - 2009 Digium,
<div class="ecxim"><br>
<br>
>> What does the general section of your sip.conf look like?<br>
<br>
</div>
[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>
<div> <br>
</div>
<br>
<br>
Warren Selby wrote:
<blockquote>
<div>
<div class="h5">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="ecxgmail_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="ecxgmail_quote" style="padding-left: 1ex;">
<div>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>
<br>
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<pre><hr size="4" width="90%"><div class="ecxim">
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<pre><hr size="4" width="90%">
No virus found in this incoming message.
Checked by AVG - <a moz-do-not-send="true" href="http://www.avg.com">www.avg.com</a>
Version: 9.0.698 / Virus Database: 270.14.39/2469 - Release Date: 10/30/09 00:52:00
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<pre><hr size="4" width="90%">
No virus found in this incoming message.
Checked by AVG - <a moz-do-not-send="true"
class="ecxmoz-txt-link-abbreviated" href="http://www.avg.com">www.avg.com</a>
Version: 9.0.698 / Virus Database: 270.14.39/2469 - Release Date: 10/30/09 00:52:00
</pre>
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<pre wrap="">
<hr size="4" width="90%">
No virus found in this incoming message.
Checked by AVG - <a class="moz-txt-link-abbreviated" href="http://www.avg.com">www.avg.com</a>
Version: 9.0.698 / Virus Database: 270.14.39/2469 - Release Date: 10/30/09 00:52:00
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