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When creating an inbound route to an extension using Destinations in Fusion, it used to create 4 lines of XML below the basic info(as viewed in Inbound Routes) similar to this: condition destination_number ^\+(12223334444)$ 20 true...

Since my last post I had to revert to this string: ^011(\d{9,17})$ Attached is a screenshot of what works for my setup. I use the same outbound gateway for International and local calls.

Apparently I wrote too soon, what I put above started to not work soon after. Apparently the phones I am using for this (Grandstream) decide to remove the 011 if dialed out of history. So to get this to work with the Grandstreams anyway the dialplan has to work with 011 or without it. After...

I had a similar problem with Twilio. Looks like they don't want the 011 coming through, only the country code and number. So I got this dial plan to work: ^011(\d{9,17})$ Not sure, but I think what it does is require the user to dial 011 (for a match for International), but does not pass it on...

I have been unable to get the "fix" to work on recent versions of either 4.4 or 4.5 -- have there been some recent changes? Perhaps this is SIP provider dependent? I use Twilio at present and they do not allow outgoing callerID that is not verified. So any remote call from a non whitelisted...

If you haven't already you'll need to set up the inbound Twilio gateways from these IP addresses: https://www.twilio.com/docs/sip-trunking/ip-addresses There are 4 east coast and 4 west coast. Calls can come through any of them, so you need to set them all up (and/or International as well)...

I looked at 3CX/Elastix stuff some time ago and noted that it is proprietary software. The costs can go up at any time, and you don't have much choice with it being proprietary. Other than the 3 year free bait, I can't see how 3cx is cheaper unless maybe for fairly small PBX's. Also used...

Been evaluating an X3 and X4 in the last year with Fusion. Never had any lockups, latest firmwares. Compared with other phones under evaluation from Grandstream and Htek, only issues with the Fanvils has been with screen glare compared to the others, no matter the adjustments. I'd recommend them!

FYI if it helps anyone else been working with Grandstream on this issue. After numerous packet captures and beta firmwares, Grandstream has fixed this issue for the GXP1620/1625 series. The fix is in the latest firmware Release_GXP16xx_1.0.4.152 Still having issues with the GRP series, hope...

Another common problem is not configuring or mis-configuring Advanced-Access Controls. Usually you whitelist all the external IP's of your SIP provider, etc. in "domains", and your LAN subnet/domain in "lan" https://docs.fusionpbx.com/en/latest/advanced/access_controls.html Again, don't know...

Have you tried removing the -nonat from /etc/default/freeswitch ? Without this in my testing I could get it to work one way or another, but something was always broken on one network or another (one way audio, calls drop at between about 30 and 90 seconds, etc.). You may not notice it...

Trying to limit the number of outbound rings for a fax gateway. Basically if someone misdials a fax, it appears the system will call 5+ times ringing around 15 times each. I'd like to limit it to around 4-5 rings so as to not drive someone crazy. I don't know this is limited to just fax as...

A workaround that seems to be good for my 4.4.10 install is to edit the fax_files.php and replace the //get fax extension section with the one from 4.5.10 Adrian showed above, again here for convenience (around line 49): //get fax extension if (is_uuid($_GET["id"])) { $fax_uuid =...

Having an issue where callers to a Grandstream phone have the 1st second of audio cut off. Test methodology: Call a Grandstream phone (I am testing models GXP1625, GRP2614). Immediatedly have the person on the Grandstream phone start counting verbally and quickly. Usually the 1 and maybe the...

These are the settings that fixed this problem for me (also using pfSense): 1. Edit /etc/default/freeswitch from the CLI and remove -nonat 2. Advanced-Sip Profiles - External (NOT Internal), ext_rtp_ip & ext_sip_ip change to autonat:XXX.XXX.XXX.XXX with the x's being the public IP 3. Restart...

Interesting . . . I've been testing FusionPBX/Freeswitch in a mid/large network and have had all kinds of NAT issues I just solved using 1.8.5 and some settings changes. My network test involved 3 networks which is probably typical for a mid sized company: External (fixed public IP, NAT'd)...