I’d love to see the Whitelist Module (FREEPBX-14488) added.
I do not care at all about backup and restore. I run all my instances of FreePBX in a Virtual Machine and I get the backup and restore through that.
I’d love to see the Whitelist Module (FREEPBX-14488) added.
I do not care at all about backup and restore. I run all my instances of FreePBX in a Virtual Machine and I get the backup and restore through that.
There are already a ton of free mobile clients for FreePBX. Grandstream’s GS Wave works great. So does Zoiper, and CSipSimple.
Bria for iOS supposedly works well, but you have to pay for it.
I would like to see an IVVR(Interactive video & voice response) app. It would be amazing if freepbx would come with an integrated menu that can not only be heard but could also be seen an controled with buttons and even forms to receive input and redirect to modules according to the input received.
A client has asked if its possible to increase a Hold timeout while the call is in transfer mode. If caller is held longer than a specific time, in transfer mode, it calls back to the originating extension, after hanging up with destination and before being transferred.
Now that the patent has expired, call recording in MP3 to save space. If that is not possible then in some other open source format.
The ability for the client to access call recording without having to give them access to the entire admin interface.
The ability for the client to create their own holidays and hours without having to access admin interface.
This is already in Sangoma 7 distro
Tweet or reach out to the dsiprouter creator? Maybe he can get you over the last leg?
Well that didn’t work again.
Same results phones that were setup 5060 ChanSIP try to register on 5060 but for some reason asterisk is throwing PJSIP registrations at the phone and no endpoint is the error.
I’ve got a FreePBX box running here at home that powers my small business’s phones. It’s been working great but I’ve got to pull the server it’s running on down soon, well I hope anyways. We’re hopefully moving (home based business) and I don’t foresee having time to get it online somewhere else, then move it again, etc.
I’ve also been thinking of moving this particular server off-site due to cost. I believe it’d be more cost effective to just run the server at a datacenter vs powering it at home, but I digress.
My questions are: Are there any special circumstances to be concerned with regarding running FreePBX on say Digital ocean or Vultr (I’m installing on Vultr right now because I couldn’t find a way to upload an ISO to DO)
Security is a concern being the server is directly on the web (I’d assume, never picked apart a data center before) Should I be worried about security for the phones or security for the web administration and user control panel? What would be a good way to be secure about this hosting?
Any suggestions or guidance?
Edit: Think a small system (think 3 lines and a single truck and maybe 2 calls at once) would work ok on a 1GB VM? Should I spring for the higher plan?
Thanks
Hi,
our suggestions that come from customers:
We made a voting for our customers in middle east. I’ll share the result with you.
Report a bug.
Allright, so in the config edit i’ll dit the ‘extensions_custom.conf’:
I want to use 900 to call the intercom
[door-intercom]
exten => 900,1,Dial Local/220www1212
exten => 900,1,Hangup
But then…how can I get Freepbx with Custom and/or Miscellaneous functions to add extension 900, and point it to the code?
Cheers
Thanks I’ll try that…
I see this is a “feature request” https://issues.freepbx.org/browse/FREEPBX-16556
OK… The issue seems to be with the email field, if i create a contact and leave the email field blank, i am able to save the contact. Once i add any text in the email field, even when i modify a already saved contact i get the mentioned error.
Good morning,
I’m discreetely new to the world of FreePBX. I’ve been able to install it on a HP server armed with esxi v6 and getting it working.
I need to achieve tho these points:
Differentiating ring tone for internal calls;
Differentiating ring tone for external calls transfered from one extension to another one;
Enable call transfered call pickup (for instance, if I transfer one call from extension 201 to 206, I want to be able to take it back in case 206 is not avaliable or user is away from his desk).
Are these options available? If yes, how can I implement those on my freepbx?
Thank you in advance,
Regards
Giovanni
Hi, I am trying to get the following setup, with follow-me.
I have a trunk, which has an incoming call, with the incoming caller ID being displayed as 0346-123456 instead of 31346123456 which is required by my trunk provider to display the caller ID outside of the call.
dit: sip:0346123456@xxxx bij from moet 31346123456 zijn. Dan zou het goed moeten gaan.
Je displayname staat wel als 31346744040 alleen je number niet.
So the question is, how do I change the incoming displayed / registred callerID number from 0346123456 to 31346123456 in Asterisk / FreePBX?
sip:31346123456@xxxx
Kind regards
Hello FreePBX,
When we login to a FreePBX, we will get a beautiful Dashboard with a FreePBX statistics window.
But in that windows there are no horizontal information about Time.
Is there a specific reason for this?
In my honest opinion, this looks great but is totally useless. I can tell that there where calls somewhere within the last hour.
Only when we hover over the diagram, we will get time information.
Would it be nice if we login and immediatly see the time-frame information on the horizontal Statistics line?
First of all, you are running a very old version of FreePBX.
You can set alert info in your inbound routes which will change the ringtone of external calls, this video will probably very helpful for you.
I am not sure how to accomplish distinctive ringing for transferred calls, you’ll likely need to add some context and custom code.
You can dial **EXT to pickup any ringing extension. Or you can setup a Call/Pickup group with specific extensions.
I actually know about this method. I can’t use it tho in this scenario as I need to pickup a call that I transfered to another extension. In this case, I receive “declined” after I dial **EXT. It actually works on extensions that are out of the queue whenever I pickup an external call. All the extensions are in the same pickupgroup tho.
Do I need to use a ring group instead of a queue?
I’ve tried to add bellcore-dr3 to the alert-info field inside the inbound route, but this way, the external call ring tone is changed, while I need to change it for internal calls.
I think so, but I’ve genuinely no idea on what to do about this.
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