Also be aware I see a lot of issues with Verizon FIOS delivered service. FIOS modem/routers are getting impossible to turn of sip alg. This usually means I have to consider a different sbc that is known to work around these type of issues. ============================ Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431
Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ ----- Original Message ----- From: Tony Graziano <[email protected]> To: [email protected] <[email protected]>; [email protected] <[email protected]> Cc: [email protected] <[email protected]> Sent: Sun Apr 25 16:46:09 2010 Subject: Re: [sipx-users] Polycom IP 335 Remote Worker Setup Yup. You might investigate whether there are any open source options fir that router (essentially replacing it with a linux OS). You might also search to see if there is a way to disable the spi on it. I'm getting ready to cross that bridge (remote user support without an external sbc). The problem as I see it are consumer grade routers that are inflexible. The models are always changing, so the problems are always "new". If you are supporting "multiple" users on a single site, you are better off with something that supports "tomato" firmware. I'm a thinking I'll have to resort to a remote pfsense box running siproxd for the remote users. I'm hoping that the local (remote branch calls across the hall) might be handled by siproxd and never "leave", while still providing presence. I have a way to go to see how that would work, then I would need to get a low power appliance to make it idiot proof. ============================ Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ ----- Original Message ----- From: [email protected] <[email protected]> To: [email protected] <[email protected]> Cc: [email protected] <[email protected]> Sent: Sun Apr 25 16:31:39 2010 Subject: Re: [sipx-users] Polycom IP 335 Remote Worker Setup Well I found out the fix and it was a combination of a few things. SIP ALG was enabled SPI was enabled along with DoS prevention and block anonymous request. I turned off all the settings and it was still happening but what I found out is that SPI scans UDP traffic and UDP traffic will loose packets if there's something in it's path. once I changed to TCP ONLY as my protocol for the phones the problem went away. I still have very short registration times but it re-registers over itself and does not have an expired registration anymore. Thanks for all your help -----Original Message----- From: "Scott Lawrence" [[email protected]] Date: 04/25/2010 04:24 PM To: "Jermaine Pinder" <[email protected]> CC: [email protected] Subject: Re: [sipx-users] Polycom IP 335 Remote Worker Setup On Sun, 2010-04-25 at 09:08 -0400, Jermaine Pinder wrote: > I’m currently testing the new IP 335 HD From Polycom with Firmware > 3.2.1 and I’m having an issue with very low register time causing the > phone to expire in 280 seconds and re-register in 60 seconds (default) > for remote workers. Are you sure this is only for remote workers? Does it do the same short registrations when on the LAN? > The problem I’m facing is the phones will only ring when there’s more > than 100 seconds left to re-resister, honestly! I wonder who thought that was a good idea? > I can call out and call any extension on my LAN and WAN side but > sometimes the WAN side phones does not ring and I checked and the > problem is the same… if there’s 100 or below seconds left then I can’t > call the extension, I have to wait until it re-register before I can > call it. > > I tried to set the expire time to 3600 or leave it blank but it still > happens. > > Here are my settings :- > > Firewall= Pfsense (works great!) > RTP ports open= 30000-31000 UDP/TCP > SIP Ports open = 5060 UDP/TCP > Signaling Ports = 5080 > Nat turned on in all the right places > > My phones registers perfectly and I can make call in/out except when > there’s 100 or less seconds on the phone. > > My Polycom Phone settings: > > On Netowrk > RPT > NAT my starting media port is 30000 > On SIP > Outbound Proxy is set correctly pointing to the pfsense external > IP port 5060 > On Lines > Server 1 address points to my internal sipx.sipdomain.com Take a trace of the registrations. Compare the 'expires' time returned from the sipXregistrar (as seen in the logs or on the wire between sipXecs and the pfsense) to what the phone receives. It may be that the pfsense is artificially shortening the registration as a NAT keepalive mechanism (some systems do this - sipXecs does not). _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/ _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
