[Asterisk-Users] Configuring Polycom 501 IP phones via the console

[Kevin Smith]

Hi Stephen,

Sorry if the e-mail is a bit choppy but I figured it would be best to 
cut/paste answers in. Now again, I am using the 601's so things may be a 
little different, but for the most part should be similar.

> No NAT. This is just one Polycom 501 that is dialing out through an
> Asterisk server with a TDM-400 card in it.
>
> I'm not using a bootserver; I figured that with one phone, I ought to be
> able to just do it locally on the phone. The impression I am getting is
> that Polycom really doesn't want people configuring the phones that way.
> The Admin guide contains slightly more than *no* information on how to
> do that.
>
> It just seems like I should be able to enter a few things on the on the
> phone console and have it working, then fine tune things for larger
> deployments later. I just want to see the thing work first.
>   
I wonder if you are looking at a different guide. The Administrator 
guide I have (in Section 2.2.2) has a whole list of advantages for using 
a bootserver. If you are going to use FTP, then you need to make sure 
the phone has the proper information to access, same with HTTP. Then you 
just need the proper files up on the location. True, for 1 phone it 
isn't needed, but I am managing about 20 phones (some in different 
states and soon more) so it is very handy to have.

> That's the trouble. So many places to configure!
Yes, I know, it took me about two days to get things finally sorted out, 
but once you get there...you will be like DUH!
> (Only one line configured for the Polycom in sip.conf, like so:
>
> [general]
> context=default
> srvlookup=yes
>
> [polycom]
> type=friend
> secret=welcome
> qualify=500             ;qualify peer is no more than 500 ms away
> nat=no                  ;this phone is not natted
> host=dynamic            ;this device registers with us
> canreinvite=no          ;Asterisk by default tries to redirect
> context=internal        ;the internal context controls what we can do
>   
Okay, above looks fine. Now here may be some confusion. The sip entry 
isn't for a line...it is just a registration for Asterisk. The 601 for 
example, one key (which you will see later) can handle 24 calls (which 
is its max), The 501 can handle 3. But this just verifies the phone has 
access to the server, the context it belongs to, etc, the number of 
lines it can use  is based on the phone and  the available channels on 
Asterisk.

> Address: [this is supposed to be the DNS or IP address of the SIP server]
> Port: 5060
> DNS Lookup: UDP only [I set this to UDP only because the internal DNS
> server we're using here only does UDP]
> Register: Yes
>   
Address is the address of the SIP server.
Port: 5060 which is default
For DNS, if you can only use UDP that is fine., and of course you want 
the phone to register.

> Now I have to set up the lines, so I go back up a level and down into
> "Line 1: ..." where I see
>
> Display Name: [don't know what this is for]
>   
Display name, is caller ID basically. If you have support for caller ID 
name, that is what it is. I do fill it in, like for example my company's 
name is on my phone config, but I don't see any reason why you can't 
leave it blank. I was thinking ahead for if/when we do SS7 or something 
the name will show up.
> Address: [what goes here? SIP server address again?]
>   
This is a little confusing, but this is the number or extension. For 
example, a phone number. You also can dial Internet addresses so that is 
why it is called an address. I believe this is also used later... but 
for now, I would set this to your extension, even if it isn't used, it 
is there for when it is.
> Label: [and here?]
>   
One the phone, next to the line keys, this will be the label..such as 
Line 1, or My Phone, it will show up there.
> Type: Private [the other option is "Shared"]
>   
I leave it at Private
> Third Party Name: [and what's this?]
>   
According to Polycom, this field must match the registration address 
value of the other registration which makes up the bridge line...what 
did I do with it? I left it blank.
> Auth User ID: polycom [here's where I assumed I had to put the extension
> name]
>   
Yes, however, again I use our phone numbers both in address and 
here...why? Because it was much easier to code in my opinion. I think if 
you leave this blank, it will use the address, but I'm not sure, which 
is why I matched it. Since polycom is your name in SIP you will want 
that there.
> Auth Password: **** [here's where I put the password "welcome"]
>   
Yes
> Num Line Keys: [left this blank]
> Calls Per Line Key: [left this blank]
>   
Here is what I was talking about earlier. Num Line Keys, is how many 
keys for numbers. For example, if you set it to 2. On the right of the 
LCD screen you will see a graphic of a phone in spots 1 and 2 and your 
contacts (if any) would follow. For starters I would set both to 1. Now, 
if you change calls per line key to 2, then it is like you have call 
waiting. You will be on a call and you will hear a beep and see on the 
phone someone else is calling.

> After making those changes, I restart the phone.
>
> With Asterisk running verbosely, I never actually see the Polycom
> register. Not surprisingly, I can't make any calls at all.
>
> The phone is getting network information via DHCP. It does get an IP
> address and even configures the DNS right.
>
> (did you use the Polycom SIP admin guide to figure out how to set up
> your 601?)
>
> -Stephen-
>   
Hopefully this will help a little more. I personally ended up just editing the configuration files. It was a lot easier in my opinion. I would get a copy of the sip.cfg and phone1.cfg files for your phone and glance through them. You can see how things are structured a little better. 

Kevin