Polycom 430 and 550 as remote station
I want to start experimenting with Polycom phones as remote stations. I have been successful with all manner of Softphone and the Snom 320's I use.
I have searched far and wide but am unable to find any info on Polycom's as remote stations. Can anyone lend a hand? I am going to try a PBXes and Teliax setup.
If I am setting up a Polycom to my own system... More of a firewall issue. An ITSP? not so much fun. Just so I knew I wasn't bonkers, I remote connected to a snom 320 and had it working in a minute.
So, while I like Polycom well enough, using an ITSP with one seems a considerable effort.
Who's your ITSP? We interop with a wide variety of both Hosted and IP-PBX platforms as listed in the URL below. I'm office based, but a few of my colleagues use ITSPs with no problems.
http://www.polycom.com/usa/en/partners_alliances/voip_interoperab...
Mitch
I didn't see an ITSP in that list for a single phone system. Aptela maybe.
There are no good general guides. I'll keep trying.
This is with Teliax. I have set up a Snom 320 and Linksys WIP330 with Teliax on another account.
I typically don't set things up this way. Remote stations back to my own gear is tough too with Polycom's.
I'd love to have a general formula.
OK.. here's my howto..
On the Trixbox:
Open up the following ports on the firewall and redirect them to your trixbox:
69/udp
5060 / tcp & udp
10001-20000 udp
On the polycom phone:
Set to tftp and point at the trixbox public ip.
That's all there is to it. They work like a dream.
-Percy
And I am finding that Polycom's while arguably the best phones on the market are almost impossible to set up in this fashion. Aptela is one ITSP that it will work with because they offer ftp for the phone.
I am using Teliax, Vitelity and less.net.
It may not be possible.
I tried setting up a Soundpoint IP 500 at my house to connect directly to TelIAX thourgh my firewall (via sip). I spent a couple of evenings and never go it to work. It would be much easier if Polycom phones would support STUN or something similar. I pretty much gave up because I could find no good instructions on setting it up manually.
Basically, I tried setting it up like I would to connect to an internal server, but using the TelIAX server address, username, and password.
Then, I also set these three settings in phone1.cfg
nat.ip="a.b.c.d" (the WAN address of the firewall that the phone sits behind)
nat.mediaPortStart="5004"
nat.signalPort="5060"
and tried opening these udp ports on the firewall and pointed them to the phones IP.
5060, 5004-5008
This didn't seem to work. But hopefully this gives you a few more clues.
I've used the SoundPoints with my own local FTP server with various service providers based on Sylantro, Broadsoft and a host of others. Some platforms require user ID and password, so make sure you have those properly configured as well. Per your questions re:STUN, this and a couple of other security features are currently on the roadmap, release TBD.
Mitch
The service shouldn't need to host your config files for the polycoms to work, you can host them on any ftp, tftp, or http server and point the phone to it (by selecting the setup menu immediatly after it boots). Also, if you can't do that, you can configure them via the web interface (which is the suckiest way, because every save makes the phone reboot).
If Aptella offers ftp service, does that mean that they also configure it for you? If so, please post your config files (sans usernames/passwords) so that we can get a clue for connecting to other ITSPs.
As I see it, the problem has to do with getting your polycom configs correct for it to work with the service (i.e. TelIAX), and be able to work through a natted firewall or dmz.
OK, here is the (mine anyway) whole scenario, I have someone whom is remote on a comcast modem. We pay for remote worker Internet and Phone. This position is going to move around a bit from house to house as this school year progresses. So, having great luck with every softphone plus Snom and Linksys handsets connecting directly to the ITSP I thought an IP handset would be the ticket.
The Snom handset is working well. Not as glossy as Polycom. Sounds almost as good.
Could I set up an ftp server? Yes, not the point. Aptela is one ITSP that offers ftp... To support Cisco and Polycom handsets directly.
When I can setup the Polycom to Teliax, then I'll swap it with the user, get my Snom back.
So, it seems that the reason that I wasn't connecting before is because I mistyped my username into my config files. Here is what you need to do to connect to TelIAX. I am putting detailed instructions not to be annoying, but just in case there are others who would like the detail.
First choose how to configure your phone - Method 1 or Method 2
Method 1. Edit files config files directly and set the phone to download the files. If you do this, you will need an ftp, tftp, or http server accessible from the web or on the local lan. I personally use ftp, with username/password authentication. You will have to set this up anyway if you ever want to update your firmware, so I'd assume this to be the preferred method. Upload a pristine copy of sip.cfg, phone1.cfg, 000000000000.cfg (if you want to upgrade the software, then also get a copy of sip.ld, sip.ver, and bootrom.ld). Also, create a text file called {macaddress}-phone.cfg (example: 0004f202316a-phone.cfg) that has your phones mac address in the name. In that file put in this text (substitue anything in {} with your appropriate data). Also, I am listing the colorado server, cause that is the one my login is assigned to, you might want to change that to your appropriate server. And the gmtoffset is set for PDT.
<?xml version="1.0" standalone="yes"?>
<PHONE_CONFIG>
<OVERRIDES
reg.1.label="TelIAX"
reg.1.auth.userId="{TeliaxUsername}"
reg.1.auth.password="{TeliaxPassword}"
reg.1.address="{TeliaxUsername}"
reg.1.displayName="{TeliaxUsername}"
reg.1.callsPerLineKey="3"
reg.1.server.1.expires.lineSeize="30"
reg.1.server.1.register="1"
reg.1.server.1.address="voip-co2.teliax.com"
reg.1.server.1.port="5060"
voIpProt.server.1.address="voip-co2.teliax.com"
voIpProt.local.port="5060"
nat.signalPort="5060"
nat.ip="{FirewallWanIp}"
nat.mediaPortStart="5004"
voice.codecPref.G729AB="1"
voice.codecPref.G711A="3"
voice.codecPref.G711Mu="2"
tcpIpApp.sntp.gmtOffset="-28800"
tcpIpApp.sntp.address="time.zivva.com"/>(after doing a preview, it appears the code tag works like crap on this BB, so you will have to delete the br tags at the end of each line in the xml above)
The next thing to do is to open udp ports 5060 and 5004-5008 and point them to your phones ip address.
When you boot the phone, hit the setup softkey and type in the password (default is 456). Edit the server menu and point to your ftp server with the appropriate username and password. Also, edit the phones IP address so that it has a static LAN address. Exit until it asks you to 'save and reboot'. When it reboots, it will look to your ftp for the proper config files and download them.
Method 2. Edit through the web interface. This is the way that you would have to do the configuration if you don't have an ftp server at your disposal. After booting the phone wait about 5 minutes then point your web browser to the phone's IP address. You can log in here with the default username/password which is Polycom/456 (Capital P!). Look through the various pages and find the fields that correspond to the the settings listed above. If you need more details, I can give you my number or email and we can talk directly, Robert (just send me a PM if you want that - I'm in Seattle, I see you're up near Bellingham). Like I was saying, this is the most annoying web interface because every time you try to save changes, the phone will automatically reboot, and you won't be able to get into the interface again for another 5min.
Yeah, once the settings are in, you can shut down the ftp server and the settings won't change. From this point on you can tweak it via the web interface.
One thing that I have noticed is that the boot takes considerably longer if it cannot contact the ftp server that it is configured with, so you might want to take the ftp server setting out again once you don't need it anymore.
I don't know for sure if there is a way around opening up firewall ports. I have always assumed that until this phone gets stun, firewall ports will have to be opened and pointed to the phone. But I have never actually tried. I will play with this later and get back.
Well I tested it and at this point it looks like you need to have ports open on the firewall.
If you don't open the SIP port (5060) your phone will just sit there churning away when you attempt to dial out. But it will not complete the call.
If you open SIP up, but don't open RTP ports (5004-8) you will get a fine outgoing call, but when a call comes in and you pick up, none of the audio will pass.
-------------
BTW: I found this in Aptela's KB:
What network ports and addresses are used by the Aptela network?
For the Polycom phone
Open ports 5060-5061 UDP and 10000-20000 UDP
Aptela Server Networks
69.25.47.128/27
66.150.122.0/24
When you say "open up" where is it being forwarded to on your local LAN? You can't just open ports up. It is quite unusual to open ports for phones, ports are usually opened up for the PBX. But as you said "open up", where do you point it to? I have used several phones remotely without opening anything up on the LAN where they reside and have not had issues with calls, I wonder why Polycoms behave differently. I have not used Polycoms yet but will do in the very nearest future since we have a new client that needs resolution with the same issues. If it persists, we may just set them up with a local TB and forget about remoting with Aptela...hosted is not so much of a great idea anyway especially if the connection goes down and all functionality including intra-office is completely lost.
One question about the Polycom web interface; Can you back up the settings and upload into another Polycom phone probably of same model and just change the IP and extension info?
@techieg
Forward the ports to the phone's static LAN address.
Here is more than enough info on SIP and NAT, why it's a problem, and discusses many workarounds and solutions:
http://www.voip-info.org/wiki-NAT+and+VOIP
more info here:
http://www.fridu.org/index.php?option=com_content&task=category&s...
The SIP phones that you have used probably have a STUN client that talks to a public STUN server as an intermediary. This makes them more NAT friendly. Polycom doesn't have this built into their phones (yet).
Before you use a Polycom, test one out and point it at an ftp server. You'll notice that any time you change the configuration through the web interface, it will write those settings to a file on the ftp server that is labeled with the phones mac address. When you hook up a new Polycom phone, all you have to do is make a copy of that file and change the name to the new phone's mac address.
This is a pretty simplistic view, but if you want more indepth info, you might want to download and read the Polycom SIP Admin guide: http://testlab.inin.com/compatibilityfiles_external/production/do...
@warmbowski
There is no logic with this. Forward a port number to a phone? What about other phones? How many phones do you want to forward the same port 5060 to? Switching and routing does not work the way you are describing it. Its just not a workable solution at all. You are gtting it all wrong...ports are only forwarded to the TB PBX.
If you have an Asterisk box of some sort on your LAN, then your forwarded (opened up, unblocked and pointed at your internal IP) ports go to that box. Having a SIP phone on the same LAN and registering to an ITSP out in the wild is not going to work on that LAN since the needed ports are pointing towards a different device (the Asterisk box.)
Now, if you have no TB or other VoIP device on your LAN and you want to connect a Polycom to an ITSP you may need to allow 5060 UDP to be forwarded to the IP of the Polycom Phone for it to function.
I was under the assumption, based on Roberts posting about " the whole senario" above, where he described that he was talkin about 1 phone at a remote location behind a nat.
But, if you were to set up another phone at this same remote location, you would assign the second phone a different SIP port in sip.conf (say, 5061), and then configure the phone to use that port for it's registration with the server, and then forward that port to the second phone.
C'est tout.
Like I said, this is not a workable solution and definitely not what what VoIP is designed to look like. This is an unusual setup, there is probably a bad switching and routing setup at the TB network or a band router, which I am sure may not be resolved by this odd method. I don't have the problems a lot of people face with NAT and SIP because I come from a switching and routing background into VoIP so I know what it entails as long as you have good/working networking equipment. A lot of people here and in general need to go back and learn a lot of switching and routing first before even stepping into VoIP and its protocols.


Member Since:
2006-05-31