[asterisk-users] setting up phones

[Danny Nicholas]

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#.

 

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From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ott Rose
Sent: Monday, July 13, 2009 12:49 PM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] setting up phones

 

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.

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From: danny at debsinc.com
To: asterisk-users at lists.digium.com
Date: Mon, 13 Jul 2009 12:12:16 -0500
Subject: Re: [asterisk-users] setting up phones

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.

 

-          exten => 99,1,Playback(tt-monkeys)

-          exten => 99,2,Playback(vm-goodbye)

-          exten => 99,3,hangup

 

When you get your phone where it can dial 99 and get a message, you will be
ready to proceed with P2P talking.

 

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From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ott Rose
Sent: Monday, July 13, 2009 12:02 PM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] setting up phones

 

Ok here is what i did. 

reinstalled asterisk (i used the make samples option) and asterisk-gui

in the gui i did the following
 created a dial plans using the defaults. no outgoing dial plans just local
 crated two users
 logged into the web interface with each phone and pointed them to our
asterisk server. Just the Proxy server and Registrar server. 

 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. 


I am just trying to use the gui like you suggestd

> Date: Fri, 10 Jul 2009 14:22:25 -0700
> From: asterisk.org at sedwards.com
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] setting up phones
> 
> On Fri, 10 Jul 2009, Ott Rose wrote:
> 
> > I don't think the GUI is editing the conf files correctly. I am not sure

> > I have configure things right. At this point i think i am going to start

> > from scratch.
> 
> Yea!
> -- 
> Thanks in advance,
> -------------------------------------------------------------------------
> Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
> 
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