[asterisk-users] setting up phones

Fri Jul 10 11:38:04 CDT 2009

Here is my conf files.

sip.conf
[general]
context=default
port=5060 ; UDP port for Asterisk
bindaddr=10.0.0.52; If we want to specify only an IP (if a computer has three different IPs) 0.0.0.0 means any IP
srvlookup=yes ; Enable DNS SRV server

[500]
type=peer
context=phones
host=dynamic
fromuser=500
secret=500
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 => 500:500 at 10.0.0.52/500
defaultip=10.0.0.60
mailbox=1001
disallow=all
allow=alaw

[501]
type=peer
context=phones
host=dynamic
fromuser=501
secret=501
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 => 501:501 at 10.0.0.52/501
defaultip=10.0.0.46
mailbox=1002
disallow=all
allow=alaw

======================================================================
users.conf
[501]
username = 501
transfer = yes
mailbox = 501
call-limit = 100
type = peer
fullname = 501
registersip = yes
host = dynamic
callgroup = 1
type = peer
context = DLPN_DialPlan1
cid_number = 501
hasvoicemail = no
vmsecret =
email =
threewaycalling = no
hasdirectory = no
callwaiting = no
hasmanager = no
hasagent = no
hassip = yes
hasiax = no
secret = 501
nat = yes
canreinvite = no
dtmfmode = rfc2833
insecure = no
pickupgroup = 1
disallow = all
allow = ulaw,gsm
macaddress = 00085d10927f
autoprov = yes
label = 501
linenumber = 1
LINEKEYS = 1

[500]
username = 500
transfer = yes
mailbox = 500
call-limit = 100
type = peer
fullname = 500
registersip = yes
host = dynamic
callgroup = 1
type = peer
context = DLPN_DialPlan1
cid_number = 500
hasvoicemail = no
vmsecret =
email =
threewaycalling = no
hasdirectory = no
callwaiting = no
hasmanager = no
hasagent = no
hassip = yes
hasiax = no
secret = 500
nat = yes
canreinvite = no
dtmfmode = rfc2833
insecure = no
pickupgroup = 1
macaddress = 00085d1095aa
autoprov = yes
label = 500
linenumber = 1
LINEKEYS = 1
disallow = all
allow = ulaw,gsm

Date: Fri, 10 Jul 2009 12:16:50 -0400
From: stotaro at first-notification.com
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] setting up phones

Change the address in sip.conf, not the phone.

On Fri, Jul 10, 2009 at 12:04 PM, Ott Rose <sixfourimpala at hotmail.com> wrote:

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. 

-- 

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Steve Totaro 

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+12024369784 (Skype)

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