Problems getting REMOTE Polycoms to register properly

[rwf]

Hi-
Using latest PIAF and have much experience with PIAF, Trixbox, AAH and other flavors and have been having pretty successful experiences so far. My server is directly on the internet in a data center and all sorts of phones and ATAs work fine.

Problem comes in this one remote office where I have Polycom Soundpoint 501 phones (8 of them). I have searched the Internet and read every message here that mentions Polycom- plus I have read Fortel's helpful howto, but so far nothing has happened except I am getting less hair.

The user has a Linksys VPN router (but is not using VPN). I don't know how it is setup because no one has the password to it and I can't risk defaulting it because it is remote to me.

All his PCs work fine, and so does ONE phone. I feel that the problem has to do with the SIP porting. His other office has all SNOM phones and they are pretty much default, with the PIAF address and credentials set properly. They all register and show in PBX info that they are on different and various high numbered ports such as 49152, 49226, etc. NAT is set to yes on the extension configurations.

However although all the phones saw the server, only one could make a call. PBX info showed them all on 5060. Fortel's howto mentioned remote phones and setting incrementing ports (5060, 5061, etc) but nothing else about remote phones. I did try putting them on incrementing ports, but it didn't help.

Problem is that there are several places to set addresses and ports in these phones. Does anyone have a successful REMOTE Polycom installation with multiple phones, and could you share info with me that might help make this work? The CEO of the customer has asked me to go onsite tomorroe and I really don't want to look like an idiot- but this problem is kicking my tail!

Those SNOMS are easy to set up, but the quality is nothing near that of the Poly.

Thanks

Ralph

 

[jroper]

Hi

This behaviour is often typical to IPCentrex setups. I do try and avoid them. The problem always lies in a combination of the phone and the NAT device - often one works and none of the others do, or they work once right after reboot and never work again.

To do this installation, you really need to control the NAT device and the phone type.

In your case, do you have an option to use a stun server within the Polycoms, if so, try setting it to stun.xten.com which is a public stun server.

You also need to look at the SIP registration in Asterisk, sip show peer 201 where 201 is the extension - with any luck the IP address it should be reporting is the public one, not the internal IP - this can help you to nail down the problem.

Joe

 

[jmullinix]

Ralph,

For what it is worth, I have had a great deal of trouble with Linksys BEFR routers, the ones with the VPN endpoint on them. For whatever reason, I just have trouble making SIP work with them.

 

[rwf]

Clarification

Need to clarify something in my first post. NAT is set to yes on the Polycom extension setups, not just the SNOMS. SNOM discussion was only to show that I do have some successful remote phones.

And when I said "However although all the phones saw the server, only one could make a call.", that was about the Polycoms.

Thanks. I hope someone can help

 

[rwf]

Yeah- I was kind of thinking that it wasn't "SIP friendly", so I was going to grab another router to substitute there to test. That of course hinges on whether the DSL login info I was told is correct or I can get the ILEC to tell me what they are.

 

[rwf]

Joe-
I am going to try all those things. Thanks for the suggestions.

 

[jwells]

Just to chime in all the remote Polycom setups I have working I am doing it via a vpn tunnel from the remote site to the main site.

I also use the ftp provisioning method in where each phone gets its dhcp info and option 66 on the remote side telling the phone where to get the config file and the ftp password. Works like a champ. now you have to have a decent DHCP server on the remote end a Linksys wont do it. If they have a Windows server you can use that as well for DHCP.


I tried to hand config the phone way myself pulled my hair out too and gave up I could get 1 to work but no more than 1. The method I now use works well takes some time to setup but again it works well with the polycom phones.

Jim

 

[mark-hc]

Jwells, where did you get the information to set all this up? I've been playing with a voip system for the past couple of weeks, but information is scattered everywhere. Most of the time I don't know if the information I'm following is outdated or what...

Should we start compiling a PIAF wiki that can stay current?

 

[wardmundy]

Helllllllo. Have you looked at the knol? You can add anything you like.

http://knol.pbxinaflash.com

The question, of course, is whether (for phone setups), it isn't just easier to search the forums.

 

[rwf]

Thanks Joe- No The Polycom does not support STUN. Many people have been pushing for it.

I pulled out some more hair at the customer's site yesterday. Updated bootrom and SIP app on all the phones and had no better luck.

I ditched the BEF router and went for the VOIP load for DDWRT v24 and I am going to try the Milkfish SIP proxy that is built in to that.

I hate to get rid of the Polycoms, because they are expensive and they work well when they work- plus the speakerphone is excellent.

So I'll continue to beat on this, but it sure is frustratng.

 

[jroper]

Hi

This is what scares me about remote phones and IP Centrex - we ought to charge 100 times as much for them, as they are at least 100 times more trouble than a PBX installation.

In the past, when I've had my fingers burnt with this, I've just gone with a PBX on site and put the cost of it down to "School Fees"

Sorry we cannot be of more help and cannot give you a magic bullet to fix this.

Joe

 

[jwells]

Ralph do you have any way to set option 66 in a dhcp server on the remote side ? and point the ftp login to the main PBX public IP ? I assume you don't have a vpn between sites

Check Chris Sherwood's site out it explains the setup as well it will help get you going
http://www.sureteq.com/asterisk/polycom.htm

Their are windows XP dhcp servers that you can run as well if you don't have any hardware capable on the remote site end


Jim

 

[Fortel]

Remote Polycoms...

I just saw the post about remote Polycoms giving you grief...

My previous write up should work in most cases. Here's what I'd do next:

Log into the phone. Click on Lines. At the Line 1 section, verify

Display Name = extension number
Address = extension number
Auth User ID = extension number
Auth Password= very strong password
Label = whatever (this will appear in display next to line key)

Server 1 = IP address (preferably, or use host name)
Port = Leave blank in this case

Expires 30
Register 1
Retry Time Out = 10
Retry Max Count = Leave blank
Line Seize Time Out = 150

Message Center

Subscriber = Leave blank
Callback mode = Contact
Callback Contact = *97

Click on Submit. Phone will reboot.

When able (it will take a few minutes,) log back in.

Click on SIP

Address = Leave blank
Port = Leave Blank
Transport = DNSnaptr

Server 1
Address = Leave blank
Port = Leave blank

Scroll down to Local Settings

Local SIP port = 5060 (This must be unique for each phone- 5061, 5062, etc)

Click on Submit. Phone will reboot. When able, log back in and click on Network.

RTP Port Settings:

Filter by IP address = Enabled
Filter by Port = Disabled
Forced Port = Leave blank
Port Range Start = Leave blank

Network Address Translation

IP Address = Leave blank
Signalling Port = Leave bank
Media Port Start = Leave blank

Click on Submit. After phone reboots, it should register- and the indication of success is a solid black telephone icon in the display next to the line key. A hollow telephone icon indicates a problem...

Hopefully, by following the above you will have success...

Peter

 

[mikemee]

Remote polycom fetching config with overrides

Just this week I stepped someone through a remote polycom config (and then a 2nd, and another coming up next week). The first one took an hour (including 30 mins of reflashing the rom, filesystem etc) and the rest stepping them through menus. Here's what worked for me the second time around (with the shortcut described below):

0) Be sure that a Polycom phone will self-configure from the pbx from inside the LAN (yes, I was using the green-system endpoint manager). Doesn't have to be the remote phone, just some Polycom to be sure the tftpboot process etc works ok.

1) Get the MAC address of the phone and create a file in /tftpboot/polycom/overrides/0004f217188c-phone.cfg (substitute your MAC obviously) containing the following:

<?xml version="1.0" standalone="yes"?>
<PHONE_CONFIG>
<OVERRIDES voIpProt.SIP.outboundProxy.address="aa.bb.cc.dd" tcpIpApp.sntp.address="us.pool.ntp.org" voIpProt.server.1.address="aa.bb.cc.dd"/>
</PHONE_CONFIG>

Substitute aa.bb.cc.dd with your pbx public IP address.

2) open the filrewall to allow outside t*f*t*p access (I used a whitelist for the end-users site because I'm paranoid)

3) step the user through a reset to defaults for local and phone settings (there are 2 separate resets)

4) step the user through Setup to change the server to t*f*t*p and then to set the public IP for the server

5) Wait while the phone rebooted and did its thing.

The config file was key the 2nd time around. The first time I had to step the user through setting the server and SIP addresses (eventually I had to remote desktop and log into the phone). Doing this allows you to

1. leverage the working LAN configuration for the phone
2. avoid messing with the phone's UI which is incredibly confusing about which server goes where (just wait until you try to set it to point to 2 pbx systems -- one per line)

Hope this helps. With this setup, the phones are very stable (two separate locations, two different stupid consumer grade routers). Prior to resetting to defaults and downloading a known working config, one of them had been having strange issues with voicemail status lights and the other had been pre-configured by hand before sending to the user (and still wasn't quite right).

cheers, michael

 

[RonRussell]

We use Polycom phones in remote (to PBX) installs frequently. But for a location that requires more than three phones we always install a remote PBX and configure an IAX trunk to the main site. This seems to better utilize the available bandwidth. Either way (remote or local PBX) works well within the limitations of the overall network. In older installs we used an iptables firewall seperate from the PBX. In new installs we are using Untangle (opensource firewall implementation). With firewall of choice you just need to create port forward rules at your PBX location to allow sip ports and either ftp or t*f*t*p ports to be accessed from your remote sites. The Polycom phones will provision at the remote site just like they do from the local site (via ftp or t*f*t*p). BTW - how much bandwidth do you have to support eight phones at the remote site?

 

[norsor]

We had similiar problems with one of our clients. They had about 35 internal extensions and 10 external extensions. We added an additional 6 remote phones and everything got real bad real quick. Phones would randomly disconnect or not connect at all until the router at the main office had been reset.

We also tried one of the Linksys VPN routers with no luck.

We ended up using a monowall firewall at the main location and it fixed the problem.

We've since placed a server at each location connected using an IAX trunk through a VPN tunnel.

 

[rwf]

Thanks, all

I ended up installing a VPN concentrator at the data center and setting up VPNs from each office, starting from the one with the Polycoms. Should have done that first, I guess!