Recent content by legolas108

OK, problem solved, but not really properly I believe. When copying the value of the global_codec_prefs variable (Advanced / Variables) to the internal SIP profile (Advanced / SIP Profiles), i.e. replacing the values (both $${global_codec_prefs}) of the inbound-codec-prefs and...

Hello! Trying to make video calls work. Followed the instructions in Video Call but the H264 codec doesn't show up under "SIP Status" as described though fs_cli -x 'load mod_h26x' outputs +OK Reloading XML -ERR [Module already loaded] Also complete machine restart didn't help. What more...

Re-installed server (Debian 10 instead of Ubuntu 18.04) and latest FreeSwitch (1.10.2, previously 1.8.4) and FusionPBX (4.5.11, previously 4.5.10) which solved the problem :-)

On internal calls while waiting for response after dialing there's the usual ring tone. But on outbound calls it has been missing for some time now and I don't recall what was changed to trigger this. Corresponding to this seems to be an error logged by FusionPBX 4.5.10 / FreeSwitch 1.8.4...

Adjusting to a dynamically changing external IP (ADSL) by maintaining a dynamically adjusted DNS entry works, but the system remains unavailable until DNS change takes effect (at least 15-20 minutes even when setting the TTL to 5 minutes). So, tried to maintain the IP directly with our VoIP...

These old systems should get fewer and fewer one would assume, so over time this issue should disappear then. Thanks so much for looking into it and giving your expert advice and so promptly, highly appreciated!

The landline endpoints are with a 20+ year old phone system, a Panasonic KX-TD1232, which accommodates analog and digital phones (and which we want to replace by the VoIP system). For the test used a Panasonic KX-TGA652 DECT 6.0 cordless digital handset.

Hi Adrian Thanks a lot for your review and background info, much appreciated. Good to know from an expert that we're (most likely) OK with our setup.

Thanks so much for your prompt response and looking into the details! Tried one other land line number and it was the same, no automatic hangup. All Twilio IPs (54.172.60.0/23, 34.203.250.0/23, 54.244.51.0/24) are whitelisted in the firewall. Logged another brief test call, complete fs_cli...

Thanks for quick response! Running FusionPBX 4.5.10 on FreeSwitch 1.8.4 in a dedicated Ubuntu 18.04 server KVM "pbx" on a Ubuntu 18.04 server box with dedicated NIC pass-through connected to the modem. There's a VPN connection from "pbx" (IP 192.168.4.3, subnet 192.168.4.0/24 dedicated to SIP...

When hanging up a call coming in to a FusionPBX/FreeSwitch SIP client from a PSTN endpoint the latter doesn't automatically hang up also. If the other endpoint is a mobile phone it will hang up shortly after as expected. What setting would make the PSTN endpoint hang up also, please?

We just tried the "Enterprise" ring group strategy and set the destinations with time offsets like this: Destination Delay Timeout 101 0 10 102 10 10 103 20 10 which emulates the "Rollover" strategy but shouldn't have a problem e.g...