[asterisk-users] setting up phones

Fri Jul 10 10:57:38 CDT 2009

I saw 127.0.0.2, never seen that before.  Loopback that I have seen is
127.0.0.1.

I always just bind to 0.0.0.0 since I have never really seen a point to
binding to a specific IP.  I guess if you are dual homed and don't want
remote phones to work, but then you could just block that stuff in IPTables
or whatever firewall.

Thanks,
Steve T

BTW, what GUI?  That was part of what I was asking when I said "what flavor
of Asterisk?"

On Fri, Jul 10, 2009 at 11:51 AM, Danny Nicholas <danny at debsinc.com> wrote:

>  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)
>  ------------------------------
>
> Windows Live™: Keep your life in sync. Check it out.<http://windowslive.com/explore?ocid=TXT_TAGLM_WL_BR_life_in_synch_062009>
>
>
>  ------------------------------
>
> Hotmail® has ever-growing storage! Don’t worry about storage limits. Check
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>
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-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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