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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;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.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in;text-indent:-.25in;mso-list:l0 level1 lfo1'><![if !supportLists]><font
size=2 color=navy face=Arial><span style='font-size:10.0pt;font-family:Arial;
color:navy'><span style='mso-list:Ignore'>-<font size=1 face="Times New Roman"><span
style='font:7.0pt "Times New Roman"'>
</span></font></span></span></font><![endif]><font size=2 color=navy
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:navy'>exten
=> 99,1,Playback(tt-monkeys)<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in;text-indent:-.25in;mso-list:l0 level1 lfo1'><![if !supportLists]><font
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style='font:7.0pt "Times New Roman"'>
</span></font></span></span></font><![endif]><font size=2 color=navy
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:navy'>exten
=> 99,2,Playback(vm-goodbye)<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in;text-indent:-.25in;mso-list:l0 level1 lfo1'><![if !supportLists]><font
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style='font:7.0pt "Times New Roman"'>
</span></font></span></span></font><![endif]><font size=2 color=navy
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:navy'>exten
=> 99,3,hangup<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:.25in'><font size=2 color=navy
face=Arial><span style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-left:.25in'><font size=2 color=navy
face=Arial><span style='font-size:10.0pt;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.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;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><o:p></o:p></p>
</div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><font size=2 face=Verdana><span
style='font-size:10.0pt;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 <st1:PersonName w:st="on">sedwards@sedwards.com</st1:PersonName>
Voice: +1-760-468-3867 PST<br>
> Newline Fax: +1-760-731-3000<br>
> <br>
> _______________________________________________<br>
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> asterisk-users mailing list<br>
> To UNSUBSCRIBE or update options visit:<br>
> http://lists.digium.com/mailman/listinfo/asterisk-users<o:p></o:p></span></font></p>
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