Hello All. I have a hosted freepbx server (FreePBX 15.0.16.22) Asterisk 13.27.1 built by mockbuild(from command line) and I’m trying to configure the baseline file in EPM for polycom to include a STUN server. Polycom documentation states that this setting is found in the firewall-nat.cfg but freebpx does not show that as a one of the files to configure. A little guidance would be appreciated.
Sorry that I don’t know the answer, but IMO configuring a STUN server on an IP phone is not a good idea, unless you have exhausted all other options. You would be depending on a third party service. If it goes down or is unreachable because of a routing issue, your phone mysteriously stops working, usually with no obvious error indication. In most cases, the STUN server is a free service run by an organization with which you have no business relationship, so no support is available.
Also, note that older Polycom phones don’t even support STUN.
If you do nothing (the phone presents its private IP address in the SIP headers and SDP), what goes wrong? What kind of router / firewall is the phone behind? Have you tried turning off any ALG? Is the phone a pjsip or a chan_sip extension? If chan_sip, confirm that you set NAT for the extension. Is the PBX listening on standard ports (pjsip on 5060, chan_sip on 5160)?
Thanks for the reply. All the extensions are pjsip on port 5060 and ALG is turned off. I have a mix of phones, cisco, polycom and grandstream spread across multiple sites. One site has two polycom phones, a VVX401 and a VVX 331. The VVX401 will register, but then will quit working after about an hour. It will not ring, if you dial a number it will take about 30 seconds for the phone to respond to your button press. I put the VVX331 in place to see if it does the same thing and it does not. ALG was what I thought the issue was initially but I have confirmed that has been turned off. I have one site with almost 20 cisco phones all behind a NAT connecting back to my freepbx with out issue. Not sure what is going on with this polycom.
Try setting the Polycom registration expiry to e.g. 120 seconds. Also try changing the local SIP Port to e.g. 5080 on the 331 and 5081 on the 401.
If no luck, report whether the Asterisk log shows the phone going unreachable, and whether the source port seen by Asterisk changes with each registration. Router / firewall make / model?
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