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
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