Recent content by pbxnerd

How about running the OpenVPN server on its own separate VPS, and using it essentially as a proxy server, so that connecting phones get that server's WAN IP. Then whitelist that WAN IP in the PBX. This allows the PBX to still be 100% locked down as far as external IPs and ports and avoids...

That's impressive @DigitalDaz . You seem to refer to this in past tense - do you still carry it out this way? And if not, why not?

I solved this by using Telnyx for SIP trunking. They publish a small set of static IP addresses for both SIP signaling and RTP media (which is uncommon). That lets you lock down your PBX to accept traffic only from Telnyx’s IPs plus your phones’ static IPs. As long as your phones are on static...

I’ll be interested to hear if anyone has it working on red hat. I’d like to do that too.

Thank you! Restarting my server seemed to clear up the issue.

I solved this by manually adding the following to my phone's configuration file in the Profile Editor: account.1.codec.opus.priority=1 account.1.codec.opus.enable=1 account.1.codec.opus.para=Opus-FB

Hi Everyone, I'm trying to provision OPUS and G722 to my Yealink T46U. G722 provisions fine, but OPUS simply does not appear. If I try to provision OPUS alone, no codecs provision to the phone at all. I have attached a screenshot of the applicable provisioning section of Default Settings...

Sure, how do I do that? I'm sorry I don't see the way to do so.

Hello everyone, I have a test Call Center Queue set up, but I have weird issues with the agent I've set up. I set it manually to log into to queue, and confirm it is available in "Call Center Agent Status". However upon calling the queue, the phone logged into the queue never rings, and the...

Thanks. I ended up switching to telnyx which allows TLS and SRTP on the external leg. Now everything works perfectly!

Thank you! I added rtp_secure_media=forbidden to the External SIP profile. After flushing cache, reloading XML, and rebooting, no difference. Without SRTP, all calls, inbound, outbound, and internal work flawlessly. With SRTP I can call between internal extensions perfectly, no errors. But...

Update: I added rtp_secure_media=mandatory to the internal SIP Profile. When I try to dial an external number, I get call failure [BEARERCAPABILITY_NOTIMPL]. However if I dial *9196 for echo test, I hear myself perfectly and see the SRTP lock symbol on my phone screen indicating session is...

Thank you so much! I really appreciate you taking the time to help me! I tried adding rtp_secure_media=optional -- and also tried -- rtp_secure_media=mandatory in the internal SIP Profile (not the external profile as my trunk provider doesn't support SRTP), but I get this error...

Thank you so much! That seems to have successfully enabled TLS for SIP! My phone can now register with TLS! I so much appreciate your help. However this does not seem to have enabled SRTP. Is there anything else I need to do for making SRTP work?

Hi, I have a working fusionPBX install, but I want to enable SRTP and TLS for SIP pn all my extensions. I haven't been able to find a complete explanation of how to do this either in the documentation or here on the forums. I'm new to FusionPBX, coming from FreePBX, so I'm sorry if this is an...