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<br>I did " set sip debug on " from the CLI<br><br>It doesn't scroll messages like it did on Fri<br><br><br>i tried 99# and the screen on the phone changed to an ip of 10.0.0.99 which isn't either one of the ips of the asterisk server. then it hung up<br><br>i do have a dial tone<br><br><br>i just figured something out after reading my post.<br><br><br>if i dial 60# it shows the ip 10.0.0.60 of the other phone then switch to the extension and the other phone rings. <br><br>still can't get the 99 to call the asterisk server to work i put in the ips of the server but it hangs up right away<br><hr id="stopSpelling">From: danny@debsinc.com<br>To: asterisk-users@lists.digium.com<br>Date: Mon, 13 Jul 2009 12:57:59 -0500<br>Subject: Re: [asterisk-users] setting up phones<br><br>
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<p class="EC_MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;">I assume you get a dial tone when you pick
up the handset? If you had a good phone-to-asterisk connection, debug would
show a connection or rejection when you did 99#.</span></font></p>
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<p class="EC_MsoNormal"><b><font size="2" face="Tahoma"><span style="font-size: 10pt; font-family: Tahoma; font-weight: bold;">From:</span></font></b><font size="2" face="Tahoma"><span style="font-size: 10pt; font-family: Tahoma;">
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style="font-weight: bold;">On Behalf Of </span></b>Ott Rose<br>
<b><span style="font-weight: bold;">Sent:</span></b> Monday, July 13, 2009 12:49
PM<br>
<b><span style="font-weight: bold;">To:</span></b>
asterisk-users@lists.digium.com<br>
<b><span style="font-weight: bold;">Subject:</span></b> Re: [asterisk-users]
setting up phones</span></font></p>
</div>
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<p class="EC_MsoNormal"><font size="2" face="Verdana"><span style="font-size: 10pt; font-family: Verdana;">added that line to the extensions.conf file because i
could find a way to add it in the GUI. I put it under the dial plan that i have
selected. i just get a busy signal i tried #99 just 99, *99 nothing works.
debugging isnt showing anything.</span></font></p>
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<p class="EC_MsoNormal" style="margin-bottom: 12pt;"><font size="2" face="Verdana"><span style="font-size: 10pt; font-family: Verdana;">From: danny@debsinc.com<br>
To: asterisk-users@lists.digium.com<br>
Date: Mon, 13 Jul 2009 12:12:16 -0500<br>
Subject: Re: [asterisk-users] setting up phones</span></font></p>
<div>
<p class="EC_MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;">Most folks (AFAIK) use TFTP to connect to
the Asterisk server. I personally use HTTP, but that took a few days of
research to figure out. You’re really only using that protocol for
configuration and log transfers. The actual lifting is done on a TCP or
UDP connection. Your posts Friday indicated that Asterisk was up and
“functional” but that you couldn’t make your phones talk to it. I’m
thinking that instead of trying to dial phone-to-phone, that you should first
make one phone talk to asterisk using this little snippet.</span></font><font size="2" face="Verdana"><span style="font-size: 10pt; font-family: Verdana;"></span></font></p>
<p class="EC_MsoNormal"><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;"> </span></font><font size="2" face="Verdana"><span style="font-size: 10pt; font-family: Verdana;"></span></font></p>
<p class="EC_MsoNormal" style="margin-left: 0.5in; text-indent: -0.25in;"><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;">-</span></font><font size="1" color="navy"><span style=""><span style="font-size: 7pt; color: navy;"> </span></span></font><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;">exten => 99,1,Playback(tt-monkeys)</span></font><font size="2" face="Verdana"><span style="font-size: 10pt; font-family: Verdana;"></span></font></p>
<p class="EC_MsoNormal" style="margin-left: 0.5in; text-indent: -0.25in;"><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;">-</span></font><font size="1" color="navy"><span style=""><span style="font-size: 7pt; color: navy;"> </span></span></font><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;">exten => 99,2,Playback(vm-goodbye)</span></font><font size="2" face="Verdana"><span style="font-size: 10pt; font-family: Verdana;"></span></font></p>
<p class="EC_MsoNormal" style="margin-left: 0.5in; text-indent: -0.25in;"><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;">-</span></font><font size="1" color="navy"><span style=""><span style="font-size: 7pt; color: navy;"> </span></span></font><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;">exten => 99,3,hangup</span></font><font size="2" face="Verdana"><span style="font-size: 10pt; font-family: Verdana;"></span></font></p>
<p class="EC_MsoNormal" style="margin-left: 0.25in;"><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;"> </span></font><font size="2" face="Verdana"><span style="font-size: 10pt; font-family: Verdana;"></span></font></p>
<p class="EC_MsoNormal" style="margin-left: 0.25in;"><font size="2" color="navy" face="Arial"><span style="font-size: 10pt; font-family: Arial; color: navy;">When you
get your phone where it can dial 99 and get a message, you will be ready to
proceed with P2P talking.</span></font><font size="2" face="Verdana"><span style="font-size: 10pt; font-family: Verdana;"></span></font></p>
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<p class="EC_MsoNormal"><b><font size="2" face="Tahoma"><span style="font-size: 10pt; font-family: Tahoma; font-weight: bold;">From:</span></font></b><font size="2" face="Tahoma"><span style="font-size: 10pt; font-family: Tahoma;">
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style="font-weight: bold;">On Behalf Of </span></b>Ott Rose<br>
<b><span style="font-weight: bold;">Sent:</span></b> Monday, July 13, 2009 12:02
PM<br>
<b><span style="font-weight: bold;">To:</span></b>
asterisk-users@lists.digium.com<br>
<b><span style="font-weight: bold;">Subject:</span></b> Re: [asterisk-users]
setting up phones</span></font><font size="2" face="Verdana"><span style="font-size: 10pt; font-family: Verdana;"></span></font></p>
</div>
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<p class="EC_MsoNormal" style="margin-bottom: 12pt;"><font size="2" face="Verdana"><span style="font-size: 10pt; font-family: Verdana;">Ok here is what i did. <br>
<br>
reinstalled asterisk (i used the make samples option) and asterisk-gui<br>
<br>
in the gui i did the following<br>
created a dial plans using the defaults. no outgoing dial plans just
local<br>
crated two users<br>
logged into the web interface with each phone and pointed them to our
asterisk server. Just the Proxy server and Registrar server. <br>
<br>
Still doesn't work. Should i be able to use the configuration server
settings form the phones web gui. it has the options for tftp, ftp, http,
https. I don't know how this is supposed to be configured. I still don't know
what the problem is and sip set debug off does display any info like it was
lastweek. <br>
<br>
<br>
I am just trying to use the gui like you suggestd<br>
<br>
> Date: Fri, 10 Jul 2009 14:22:25 -0700<br>
> From: asterisk.org@sedwards.com<br>
> To: asterisk-users@lists.digium.com<br>
> Subject: Re: [asterisk-users] setting up phones<br>
> <br>
> On Fri, 10 Jul 2009, Ott Rose wrote:<br>
> <br>
> > I don't think the GUI is editing the conf files correctly. I am not
sure <br>
> > I have configure things right. At this point i think i am going to
start <br>
> > from scratch.<br>
> <br>
> Yea!<br>
> -- <br>
> Thanks in advance,<br>
> -------------------------------------------------------------------------<br>
> Steve Edwards sedwards@sedwards.com
Voice: +1-760-468-3867 PST<br>
> Newline Fax: +1-760-731-3000<br>
> <br>
> _______________________________________________<br>
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --<br>
> <br>
> asterisk-users mailing list<br>
> To UNSUBSCRIBE or update options visit:<br>
> http://lists.digium.com/mailman/listinfo/asterisk-users</span></font></p>
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