[asterisk-users] setting up phones

Fri Jul 10 10:33:19 CDT 2009

so i filled in the info and now i get this when i run  sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
500/500                    127.0.0.1        D          5060     OK (1 ms)
501/501                    127.0.0.1        D          5060     OK (1 ms)
2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]

I still cannot call the extensions and the phones say no service on there screen

From: danny at debsinc.com
To: asterisk-users at lists.digium.com
Date: Fri, 10 Jul 2009 08:40:49 -0500
Subject: Re: [asterisk-users] setting up phones

Let’s draw this out and let you fill in
the blanks.  Your asterisk server has a name of foobar.com and an ip address of
192.168.23.1.  phone 1 has ip address of 192.168.23.2.  phone 2 has ip address of
192.168.23.3.

Sip.conf should look  this

[phone1]

type=peer

context=phones

host=dynamic

fromuser=phone1

secret=secret1

canreinvite=no

directrtpsetup=no

call-limit=3

nat=no

qualify=yes

register=no

session-timers=accept

session-expires=60

session-minse=120

session-refresher=uac

register => phone1:secret1 at foobar.com/phone1

defaultip=192.168.23.2

mailbox=1001

disallow=all

allow=alaw

[phone2]

type=peer

context=phones

host=dynamic

fromuser=phone2

secret=secret2

canreinvite=no

directrtpsetup=no

call-limit=3

nat=no

qualify=yes

register=no

session-timers=accept

session-expires=60

session-minse=120

session-refresher=uac

register => phone2:secret2 at foobar.com/phone2

defaultip=192.168.23.3

mailbox=1002

disallow=all

allow=alaw

assuming your phones are set up to contact
192.168.23.1 with username phone1/phone2 and proper secret, all should register
and you should be good to go.

From:
asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ott Rose

Sent: Friday, July 10, 2009 8:33
AM

To:
asterisk-users at lists.digium.com

Subject: Re: [asterisk-users]
setting up phones

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 

+18887771888 (Toll Free)

+12409381212 (Cell)

+12024369784 (Skype)

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