[asterisk-users] setting up phones

[Danny Nicholas]

You are running asterisk as a local service (127.0.0.1 is localhost).  You
need to use the address from ifconfig (192.168.X.X) in sip.conf (bindaddr).
This will make asterisk where your phones can "talk" to it and register.

 

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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: Friday, July 10, 2009 10:33 AM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] setting up phones

 


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

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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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