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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: cliconnect@cliconnect.com<br>To: asterisk-users@lists.digium.com<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 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 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>
</div>
</div>
<br>
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<pre><hr width="90%" size="4"><div class="ecxim"><br>_______________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a></div></pre>
<pre><hr width="90%" size="4"><br><br>No virus found in this incoming message.<br>Checked by AVG - <a href="http://www.avg.com">www.avg.com</a> <br>Version: 9.0.698 / Virus Database: 270.14.39/2469 - Release Date: 10/30/09 00:52:00<br><br> </pre>
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<pre><hr width="90%" size="4"><br>_______________________________________________<br>-- Bandwidth and Colocation Provided by <a class="ecxmoz-txt-link-freetext" href="http://www.api-digital.com">http://www.api-digital.com</a> --<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a class="ecxmoz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a></pre>
<pre><hr width="90%" size="4"><br><br>No virus found in this incoming message.<br>Checked by AVG - <a class="ecxmoz-txt-link-abbreviated" href="http://www.avg.com">www.avg.com</a> <br>Version: 9.0.698 / Virus Database: 270.14.39/2469 - Release Date: 10/30/09 00:52:00<br><br> </pre>
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