[asterisk-users] setting up phones

Fri Jul 10 10:40:36 CDT 2009

I have the GUI setup and I setup users in the gui before. I still couldn't get it to work. I don't have any SIP trunks setup via the GUI because I can't figure out my settings and I was told I didn't need it to test extensions.

I am not sure what you mean by 
"Try calling out via bandwidth with SIP verbose on and post your results.
Call the other phone and post verbose.
You do have logic in extensions.conf do you not?"

I don't know how to do that.
Date: Fri, 10 Jul 2009 11:08:59 -0400
From: stotaro at asteriskhelpdesk.com
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] setting up phones

Extension 500 is registered just fine.  200 OK

Maybe you should start with a GUI version of Asterisk.

Try calling out via bandwidth with SIP verbose on and post your results.

Call the other phone and post verbose.

You do have logic in extensions.conf do you not?

Thanks,
Steve Totaro

On Fri, Jul 10, 2009 at 10:58 AM, Ott Rose <sixfourimpala at hotmail.com> wrote:

Carrier is bandwidth.com

we are running Asterisk 1.6.1.1

i ran sip set debug on from the CLI

Once i did a module reload it started displaying all the debuging info. Here is some of the debug info

--- (13 headers 0 lines) ---
Scheduling destruction of SIP dialog '5fafc3b57e3928877141d12f58c9f7a2 at 127.0.0.2' in 32000 ms (Method: REGISTER)

[Jul 10 06:53:39] NOTICE[5641]: chan_sip.c:16397 handle_response_register: Outbound Registration: Expiry for dynamic is 120 sec (Scheduling reregistration in 105 s)

<--- SIP read from UDP://127.0.0.1:5060 --->

SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK613dcb99;received=127.0.0.1;rport=5060
From: <sip:500 at dynamic>;tag=as51c22cdd
To: <sip:500 at dynamic>;tag=as51c22cdd
Call-ID: 7e1e2c4c702c5b1619fef3961219273a at 127.0.0.2

CSeq: 117 REGISTER
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Expires: 120
Contact: <sip:500 at 127.0.0.1>;expires=120

Date: Fri, 10 Jul 2009 10:53:39 GMT
Content-Length: 0

Date: Fri, 10 Jul 2009 09:42:31 -0400
From: stotaro at asteriskhelpdesk.com

To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] setting up phones

Who is the carrier?  What flavor of Asterisk are you using?

Regardless, the phones should register and be able to call each other and other Asterisk apps if you have them in the dialplan.

If you go to the Asterisk CLI and turn on SIP debugging, do you get anything at all?

also, change registersip to yes.

Thanks,
Steve

On Fri, Jul 10, 2009 at 9:32 AM, Ott Rose <sixfourimpala at hotmail.com> wrote:

Here is my physical network.

We have a Adtran router that is plugged into the Asterisk server and into the circuit provided by my tel co. 

the other nic in the Asterisk box is plugged into your lan switch

the phones are plugged into the lan switch

I can ping the phones from the Asterisk server. 

Date: Thu, 9 Jul 2009 17:42:43 -0400
From: stotaro at asteriskhelpdesk.com

To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] setting up phones

On Thu, Jul 9, 2009 at 5:12 PM, Ott Rose <sixfourimpala at hotmail.com> wrote:

I followed it the best I could. the phones say no service. I haven't got to setting up the SIP trunk yet I was told I could get the extensions to work so I could test between the two phones i have. I have to nics in my server. one is connect to the phone router the other to a network switch. which ip should it point to? I am guess the one connected to the switch. That is the one i can access the GUI from. Below are my users.conf setting. Notice all the spaces. I didn't put them in there they are like that in the conf

Either you did not explain your network topology very well or that is your problem.

Unless you are trying to segregate your VoIP traffic, plug everything into the switch.

If using DHCP, get the IP and try pinging the phones from the Asterisk box.

I bet it is just a network issue. 

-- 
Thanks,
Steve Totaro 
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