Despite my (and others) best efforts I cant get FreePBX to work with the network config I have chosen - so I’m going to buy the pbxact 40. IS there a different forum for that device?
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I have given up :(
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A feq questions
Saw your post yesterday and had in plan to write you today, unfortunately i had a busy day and @cynjut got on the boat before me…
All i can say, i suggest you to read the Queue Wiki. Most of your configurations is made out of “guessing” the right way to route your calls. I know… you’ll have to restructure your Queue, but almost all of your questions/mistakes will be gone.
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Kamailio and Siremis with FreePBX
Of course I did, and that offers me no solution.
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I have given up :(
Same Forum.
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Chan_mobile and freepbx
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Converting recorded wav calls to mp3
Crickets
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Whoops error on checking online for module upgrade
That mostly fixed me, but now I have a Security Warning:
"Module: “FreePBX Framework”, File: “/var/www/html/admin/libraries/Composer/vendor/bin/uuid altered”
I tried “fwconsole ma refreshsignatures” but that did not get rid of the issue …
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Kamailio and Siremis with FreePBX
Go back to your softphone, set it’s ip address up as a PBX in dSIPRoputer, make sure that you have a mappng through outbound global routes to a carrier that is known to work with direct IP routing (trunks that need registration/username/passwords will have to be handled in Kamailio/Siremis) . When you get that working then move on to FreePBX
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Whoops error on checking online for module upgrade
upgrade framework
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Whoops error on checking online for module upgrade
fwconsole ma upgrade framework
returns
“framework is newer than online version, unable to upgrade”
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Whoops error on checking online for module upgrade
Do it through the web browser
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Whoops error on checking online for module upgrade
That fixed it. Thanks!
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Upgrade and Centos errors
It seems that during my recent upgrade to sng7 my system has stopped. Currently I can not activate my system and the latest version of the kernel hangs on reboot.
rpm -q kernel
kernel-3.10.0-514.10.2.el7.x86_64
kernel-3.10.0-514.26.2.el7.x86_64
kernel-3.10.0-693.11.6.el7.x86_64 (This is the one that hangs)
Should I remove that kernel to see if I can still upgrade my system, or will that do more damage?
Please advise.
Mike
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UCP Can't Originate Calls
Does anyone have UCP in FreePBX 14 completley working?
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I see now that the Phone is UCP is registering successfully over https, it rings when i have a incoming call but no caller information other than it gives me a button to answer the incoming call. When i click to pickup the call nothing happens… same when i try to place a call…
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I created a new contact group, tried adding a contact, i got an error "There was an error. See the console log for more details"
Console log:Uncaught TypeError: Cannot read property 'push' of undefined at HTMLInputElement.<anonymous> (jsphp_e062e19f8ed3a714dfa91c6efc072ce9.js?load_version=v14.0.2.1:1595) at Function.each (jquery-3.1.1.min.js?load_version=v14.0.2.1:2) at jQuery.fn.init.each (jquery-3.1.1.min.js?load_version=v14.0.2.1:2) at HTMLButtonElement.<anonymous> (jsphp_e062e19f8ed3a714dfa91c6efc072ce9.js?load_version=v14.0.2.1:1589) at HTMLButtonElement.dispatch (jquery-3.1.1.min.js?load_version=v14.0.2.1:3) at HTMLButtonElement.q.handle (jquery-3.1.1.min.js?load_version=v14.0.2.1:3) -
When forwarding a voicemail in UCP FreePBX 13 you started typing a name or an extension number and it populated a list with all matching extensions, this doesn’t happen.
I appreciate if anyone can please help out.
Thanks
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What would you like to see added in FreePBX 15?
On the Dashboard I would love to see a breakdown of what
-extensions
-Trunks
are “online” and “offline” v. 14 and prior just give a total. I want to know the names.
Also on the dashboard a gui view of all active calls from and to
-extensions internally
-and to and from outside callers
I would also like to see a dashboard timeline of faxes sent and received.
On the dashboard what are the current active
-time groups
-time conditions
-call flows
That are directing calls to ivrs, extensions, and ring groups. And show their destinations as a clickable link.
Allow the dashboard elements to be customized by location, size, and show/hide
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System error with default boot option
Look at this link it will allow you to remove the kernel I believe. I have not attempted this as of yet in fear my system will not reboot.
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What would you like to see added in FreePBX 15?
Provide interface to add Windows Server Share as a backup location.
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Chan_mobile and freepbx
ok. I know this is not a NAT or Firewall issue because this is not SIP.
Besides I also have a SIP trunk on the same Asterisk setup and it works fine. So SIP-to-SIP is fine, but SIP-to-MOBILE is not.
There is a problem preventing the audio flow. Anybody who has experience with chan_mobile could help me…
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Chan_mobile and freepbx
I set debug level 1 on Asterisk.
I get :
[2018-01-18 01:36:10] DEBUG[8288][C-00000002]: audiohook.c:272 audiohook_read_frame_both: Read factory 0x6de69efc and write factory 0x6de6a91c both fail to provide 160 samples
while ringing.
and I get :
[2018-01-18 01:36:20] DEBUG[8288][C-00000002]: audiohook.c:333 audiohook_read_frame_both: Failed to get 160 samples from write factory 0x6de6a91c
[2018-01-18 01:36:20] DEBUG[8288][C-00000002]: audiohook.c:272 audiohook_read_frame_both: Read factory 0x6de69efc and write factory 0x6de6a91c both fail to provide 160 samples
[2018-01-18 01:36:20] DEBUG[8288][C-00000002]: audiohook.c:333 audiohook_read_frame_both: Failed to get 160 samples from write factory 0x6de6a91c
[2018-01-18 01:36:20] DEBUG[8288][C-00000002]: audiohook.c:272 audiohook_read_frame_both: Read factory 0x6de69efc and write factory 0x6de6a91c both fail to provide 160 samples
[2018-01-18 01:36:20] DEBUG[8288][C-00000002]: audiohook.c:333 audiohook_read_frame_both: Failed to get 160 samples from write factory 0x6de6a91c
[2018-01-18 01:36:20] DEBUG[8288][C-00000002]: audiohook.c:272 audiohook_read_frame_both: Read factory 0x6de69efc and write factory 0x6de6a91c both fail to provide 160 samples
when the phone is answered.
could this help?
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Kamailio and Siremis with FreePBX
I managed to get dSIPRouter to send some packets to and from my SIP trunk however it just wouldn’t work properly (even with softphone)
I’m at the point of giving up now.I can’t find anywhere that explains exactly the terminology / couple of fields you need to fill in Siremis in order to create and proxy through a SIP trunk - it looks quite feature rich but the documentation is clearly written by the people that have either developed the system or already have an overwhelming knowledge of how it alll works - not for anyone that actually wants to figure out how to use it.
I’m closer to just setting up another PBX now in order to route basic SIP traffic! 
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