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I have create a lua script and I pass in it 3 parameters. Lua execute by a dialplan see image. Everything is fine but.... .... When I try to connect to remote mySQL Server its does not connect any more. It not possible to connect to database. It was working for more that 1 week... but now...

Take a look here ... I hope it helps https://www.pbxforums.com/threads/voicemail-to-e-mail.475/#post-2112

Hi, I use 4.4.1 version When I try to send fax, I get this error in fs-cli console 2018-11-19 19:40:14.393636 [ERR] mod_lua.cpp:203 ...eeswitch/scripts/resources/functions/database/luasql.lua:35: module 'luasql.postgres' not found: no field package.preload['luasql.postgres']...

Hi, We have Fusion PBX on Cloud and its working since long time. We have one gateway truck. We have registered 8 extensions. Any incoming call going to a ring group (simultaneous) which it includes all the extensions. The issue is that some times when we pick up a 4th or 5th incoming...

I can not login too. I have a user test to Domain 010test.pbxxyz.com At the past I was able to use https://010test.pbxxyz.com with username test with not issue. Now I can not login even if I reset the password of user test. Can you advise please ??

What about codecs? Do you have any issue if you make calls between extensions or the issues take place only from inbound calls (from a voip provider)? Are you sure that all parties support the negotiations codecs ?

There is somebody to give me an advise ?

Line of Logs sofia/external/+402810881389@46.4.60.4 Regex (FAIL) [402102241030] destination_number(XDHGS6F90038) =~ /^(402102241030)$/ break=on-false I need to look like this sofia/external/+402810881389@46.4.60.4 Regex (FAIL) [402102241030] destination_number(402102241030) =~...

How many Gateways you have? Make sure that in sip status you have the same number of Gateways. About the Audio make sure that you all range of RTP ports (Audio Ports) are open.

Can you advice the correct Dialplan on freeswitch so I will be able to use both options (destination_number and ${sip_to_user}) ? I tried : <action application="export" data="callee_id_number=402102241300"/> But nothing changed. See image of Freeswitch Book

I am sure that I miss something at the freeswitch dialplan.... I had the same issue when I was using Grandstream IP/PBX, which lood for calle number only in destination_number field (it did not have option to choose ${sip_to_user}).

I make a new post with better explanation https://www.pbxforums.com/threads/callee-destination-number.2627/

Hi, FusionPBX get inbound calls from Freeswitch For Inbound calls routing I need to use condition = ${sip_to_user} Me I Would like to use as condition = destination_number Right now condition = destination_number is not working for me There is someone to advise me how to build the...

I try to manipulate channel variable in dial plan xml at freeswitch site https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables

Do you know how to change the channel variables at freeswitch so I will have dialed number I destination_number variable and not I sip_to_user? I have try to export it before bridging... In dialstring but I did not have positive results.

At inbound route of fusionpbx I must put sip_to_user variable. Do you know what I have to change at my freeswitch settings so I will use the destination_number variable at inbound route?

Yes is 5080 external profile... Fusion box has default installation setting. The issue is not by the fusion pbx. I think something is not set correctly at the freeswitch side. Most probably freeswitch does not set in channel variable the correct value. Do you have something to advice?

Hi, I would like the advice in case is possible. I have a freeswitch installation. I have also fusionPBX installation that is working really fine. FusionPBX uses as gateway a SIPDEVICE of my freeswitch for inbound and outbound calls. My Question has to do with this: Inbound calls from the...

Hi, I still mak tests/checks of Fusion pbx . Fusion is on public IP on Dedicate server /Debian 8 We have Extension A (192.168.0.20) and Extension B (192.168.0.30) and both are registered to fusion pbx via NAT/Router Extension A call to Extension B and they are on an active call. When the...

Take a look here Some one from IP 69.30.212.11 send invite If I had and outbound rule for the destination 000442080892571 This person would be possible to make a call for free via pbx In case I make mistake forgive me please..... 2017-12-17 22:03:19.789711 [DEBUG]...