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

[Stephen Bosch]

Hi, Kevin:

Kevin Smith wrote:
> Hi Stephen,
> 
> I use the 601's but  I don't think they are THAT much different that
> this information won't be helpful or get you in the right direction.
> 
> What is your network setup like? Are you using NAT or does the phone
> have a public IP address? Also are you seeing any errors on the CLI of
> asterisk? I know you said your configurations are local, but are you
> using a bootserver (which can be local) to grab the files?

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.

> Things I would check if you are using NAT (I think 2-5 need to be done
> in the web interface):
> 1. Make sure your SIP.conf file is configured to use NAT and give it a
> port to signal on, say 10000 for example (which I will use below to, but
> change to better fit what you would like).
> 2. Assign the phone an internal address, add port pass thrus for UDP
> packets 10000-100050 (I think should be enough) for that IP.
> 3. Assign RTP port range to start at 10001
> 4. Make sure you have a NAT address listed in the phone and you have the
> signaling port set to 10000 and Media start port at 10001.
> 5. Also if you are using a DNS name for the server (such as
> server-1.whateva.com) I use TCPperferred for DNS lookups.
> 
> If you are not using NAT, it pretty much should work out of the box
> provided it knows where the server is going. Of course, make sure the
> SIP username and password are correct.

That's the trouble. So many places to configure!

Example from the phone console:

Menu | Settings | Advanced | <enter password> | Admin Settings | SIP
Configuration

Now you see a list of parameters:

Server: ...
Outbound Pro... ... [Outbound Proxy]
RFC2543 Hold: No
Calls Per Line K... [Calls per line key]
Line 1: ...
Line 2: ...
Line 3: ...

(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

I've tested the installation with a softphone, and it works.)

Here's one source of confusion --

The parameters in the "Server: ..." category are

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

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]
Address: [what goes here? SIP server address again?]
Label: [and here?]
Type: Private [the other option is "Shared"]
Third Party Name: [and what's this?]
Auth User ID: polycom [here's where I assumed I had to put the extension
name]
Auth Password: **** [here's where I put the password "welcome"]
Num Line Keys: [left this blank]
Calls Per Line Key: [left this blank]

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-