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Is it possible to create a dialplan or use a native app to achieve the following feature: 1. Call a destination number in FusionPBX 2. Ask to enter a predefined code (acting as a PIN) 3. Enter number to forward to (step 2 and 3 can be combined together, i.e., *5551234567890) 4. Call gets...

My global codec pref (greedy) is PCMU, PCMA, G722. Through late negotiation, I was able to achieve end-to-end G722 calls via some carriers on some external calls, which is great. But as soon as such a call is put on hold, my softphone (Bria) shows that the codec immediately changes to PCMU, and...

If I use a provider for outbound calls only – for example, Flowroute for toll-free termination, do I still to whitelist the IP addresses published by Flowroute in my firewall and access control list (assuming my firewall does not restrict outbound connection to the Internet by default)? My...

I just did a fresh installation (master branch on Debian 10). In the Time Conditions app, when creating an condition initially, the page display the correct holiday presets. But once it's saved, going back into the condition displays some skewed items. See attached screenshot. Note that...

I installed FusionPBX a few months ago using the official quick installer, right before it switched from stable (4.4) to master. Later, on IRC channel I learned that 4.5 has some important fixes, but upgrading from 4.4 to 4.5 was no easy procedure and an official guide was being worked on. I...

I have the global_codec_prefs set to PCMU,PCMA,G722. But I want to change this order to put G722 first for all phone calls between internal extensions only. At the same time, I do not want this to potentially mess up transferred calls: so external caller A calls internal extension B using PCMU...

I read the documentation here about the commands to add a number to blacklist: https://freeswitch.org/confluence/display/FREESWITCH/mod_blacklist I followed the example, here's my dialplan: <extension name="add_to_block_list" continue="false" uuid="xxxxxx"> <condition...

I'm new to FusionPBX and VoIP. Just set up my FusionPBX with Anveo Direct. Regular outbound calls by number works, but I can't make SIP calls using SIP URI such as sip:test.time@sip5060.net Can someone please kindly let me know what settings are needed to make this work?