[asterisk-users] setting up phones
Danny Nicholas
danny at debsinc.com
Fri Jul 10 11:20:29 CDT 2009
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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