[asterisk-users] setting up phones

Fri Jul 10 11:33:32 CDT 2009

yes

From: danny at debsinc.com
To: asterisk-users at lists.digium.com
Date: Fri, 10 Jul 2009 11:20:29 -0500
Subject: Re: [asterisk-users] setting up phones

Phone 1 has 500 in all of it’s id’s and
connects to server 10.0.0.52?

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 11:05
AM

To:
asterisk-users at lists.digium.com

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

Great i changed it to my ip here
is the debug and sip show peers. phones still say no service i get a dial tone
when i pick it up and a busy signal when i call the other extension. 

Name/username             
Host            Dyn Nat
ACL Port     Status

500/500                   
10.0.0.52       
D          5060    
OK (1 ms)

501/501                   
10.0.0.52       
D          5060    
OK (1 ms)

2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]

--- (12 headers 0 lines) ---

Really destroying SIP dialog '1e56b3345aabbe4373a46fb973476e72 at 10.0.0.52'
Method: OPTIONS

linux-zswk*CLI>

<--- SIP read from UDP://10.0.0.52:5060 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.0.0.52:5060;branch=z9hG4bK1210f27b;received=10.0.0.52;rport=5060

From: "asterisk" <sip:asterisk at 10.0.0.52>;tag=as66b3ded8

To: <sip:500 at 10.0.0.52>;tag=as66b3ded8

Call-ID: 56cf67f56e3cba6602965f6317656c1a at 10.0.0.52

CSeq: 102 OPTIONS

Server: Asterisk PBX 1.6.1.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces, timer

Contact: <sip:asterisk at 10.0.0.52>

Accept: application/sdp

Content-Length: 0

From: danny at debsinc.com

To: asterisk-users at lists.digium.com

Date: Fri, 10 Jul 2009 10:51:18 -0500

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

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.

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

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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