Install SIP phones on IP Office

Had a few calls from customers on SIP phones.  Obviously, if you are using the J100 series on IP Office R11, you are going to have to implement SIP phones.  J100 ONLY run SIP, when connected to IP Office R11.  HOWEVER, they do work on R10.1, but ONLY as h323 phones.  So, why would you downgrade all those phones just to run them on 10.1, when 10.1 will be going EOS with the next major release?  Just upgrade the damn IPO.

So, to get the J series working, we need some settings within IPO set, and then the IPO will autogenerate the 46xxsettings.txt file.

 

First we need to navigate to the System tab, and go into LAN1>VOIP

ENABLE SIP registrar.  This is disabled by default.

If this is cloud, or you will have remote workers, you will need to allow SIP Remote Worker.

Set the SIP Domain Name.  This is used for SIP endpoints to register to the IP Office.  This should also match the domain suffix of the SIP Registrar FQDN to the right.  In the below picture it is the example.com entry.  NOTE.  If you are going to set up resilience for IP phones, this will need to be the same on ALL systems in the network.  Also note if you are using TCP/UDP and NOT TLS, then you can leave this blank and the IPO will insert the LAN IP instead.  If using TLS, YOU MUST have a Fully Qualified Domain Name entered here.

In the SIP Registrar FQDN, This is the fully-qualified domain name for the system, for example ipoffice.example.com, to which the SIP endpoint should send its registration requests.  This address MUST be resolved by DNS (public, if applicable or private).  For Vantage and Equinox this must be set.  If not using TLS, this can be the same as the SIP Domain Name entry.  For resilience, this value, if set on the failover server, is the value passed to Vantage and Avaya Equinox clients as the address for resilience. If not set, the system’s IP address is sent to those clients as the failover address instead.

Set wether you want TCP/UDP or TLS

You can customize which port you use.  If using TCP/UDP, it is highly suggested that you use a non standard port from 5060.  Something like 5065, 5070, etc….

 

 

 

Now go down to the VoIP tab, and set which codecs you want to use.  This is up to each install wether you want a wider codec that sounds better but takes more bandwidth.

 

 

Now we can create our SIP extension, so go to Extensions, and add a new SIP Extension.  If you dont want to here, you can skip this and go to the user, if the extension does not exist, it will give a pop up (web manager only) asking if you want to create the Extension.

You can lock down the codecs used here, but if you set this in the system tab, should be good.  But you do need to set the Extension Password.

 

 

We can add a user now.  The settings you need to pay attention to are:

• Extension – This should match the SIP ID of the “Base Extension” setting.
• Login Code – This can match the “Phone Password” from above, or be the Extensions login code.

If you did not do the above, when you save this and there is no matching extension created, you will be prompted to create an extension.

 

To alleviate some issues, at this point i would DISABLE “Allow Direct Media Path”.  once you know your phones work, you can set this up, but may cause issues if the network is not set properly.

 

 

Now you can boot your phones up.  When the J series boot, if they are newer SIP firmware, they may ask if you want to perform automatic provisioning.  You can, if you have access to the DES server (des.avaya.com), which is handy, but not necessary.

So, select NO for auto provisioning.  Once the phones bot up, if we are touching the phones (for sake of ease lets do that for now, as there are previous posts on setting a DHCP server) .  When you can get into the phones admin, go to servers, and set the http or https file server to point to your IP Offices LAN/WAN IP.

The phone now requests the settings text file appropriate for its particular model from the file server. This file contains a wide range of phone settings including details of the SIP server and protocols it should use and the certificate name if using TLS.

IF you have a 46xxsettings.txt file in teh Primary folder directory, now is the time to delete it.  When a file is requested, for example the 46xxsettings.txt file, if it does not exist, it will auto create a new one, with the settings we have done up to this point.  You can browse to 192.168.42.1/46xxsettings.txt and see the file create, reflecting the FQDN, and SIP domain settings.  Also at the top, you will see “AUTOGENERATED”

So the phone should grab any upgrades and upgrade per your release.  Yo can watch Sysmon if you want to see the files transfer.

That should be good enough.  I would recommend doing baby steps, if your end install will be TLS/SRTP/Remote worker etc…if the phones can come up with these minimal setting, then we can set the certificate and then TLS, and then remote worker.