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Multiple Extensions Answering A Call?

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We've discussed this in our triage meetings as I believe there are other tickets with a similar request. What you are asking for is nothing more then a page group that does not auto-answer. In essence, a page group does this but happens to go through the part of the dialplan that adds the appropriate headers to tell a phone to auto-answer. You would eliminate that. (If you happened to have phones that are not capable of auto-answering, then you would already have this.) The other things that may need to be added from a configuration perspective is if you honor their ring times or make that settable from the page group, and properly coding the calls so they don't drop into voicemail, similar to how calls in a ring group are treated.

If you're interested in exploring having this accelerated you can contact support to discuss the possibility of funding it. I suspect it will eventually get done based on our discussions, but the timing of that is in question, and it's not clear if that is something that would be added to the normal paging or to paging pro. (The decision would depend on what's involved and if any of the paging pro capabilities were desired in the implementation).


Debian8 fresh install Error Class Userman not found

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You can try but unless someone can replicate it you're going to have problems again. It would be nice if next time you can do some troubleshooting to help us figure it out.

Backup Module Cron Exception

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Nobody any idea or the same problem? Do I need to provide something else?

Debian8 fresh install Error Class Userman not found

Debian8 fresh install Error Class Userman not found

Grandstream GXP-2140 can't rebuild config

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Thank you for the assistance nebinur. In this case the problem isn't Grandstream but is definitely a minor bug in the End Point Manager. I have a couple other minor issues, that I will go ahead and open a ticket with the Grandstream Help-Desk for.

In particular I seem to have a problem where the phone fails to pick up an IP sometimes and I need to reboot. I have deployed about 100 phones (all except for 2 Grandstream), and I get this randomly on different phones. The final MAJOR annoyance I have is that it makes all of the line keys says the extension number (and name), and if I change for example all of them but 1 when call waiting etc doesn't work!

Problem with delay in call setup

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Thank you. That was the source of the problem. Removed the stun server and the delay went away.

Astcrapper Sound Files (anti-telemarketer)


Is there a way to automatically logout particular extension?

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Hi,
I am wondering whether particular extension(s) can be automatically logout in certain time (lets say 1:00am) or after certain period of login time. Is there such configuration possible through FreePBX GUI?

Stay and away and app

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To add - you can use the Call-Forward and Find/Follow-me functionality combined with the Do Not Disturb function to get all kinds of this working.

There are lots of ways to do this. I wrote a little script that runs on the PBX server that listens on the BlueTooth "network" and when a cell phone/headset/vehicle leaves the building the office phone associated with that cell phone is forwarded to the cell phone.

On the systems where I don't want to mess with BlueTooth and I'm the admin, I just go into the Extension and set up Find/Follow-me so that if my office phone rings for 20 seconds and my cell phone rings for 30, the call goes to voice-mail on the PBX. (The cell phone voice mail kicks in at about 35 seconds).

There is also a "call forward" feature that you can use. If you have control over your button configuration, you can set one up to do a call forward to a specific number of extension. One of my customers does this for his "personal" direct line when he is going to be gone for a while and wants to have the office manager handle his calls.

Since I'm not a user and the people I support are not typically "tech savvy", I usually end up setting up this kind of stuff for them using the Admin screen.

Yealink T-32G won't provision!

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Firmware 32.70.1.33, which according to the Yealink Web site is the latest firmware.

Yes, using tftp.

My tftpboot directory has the .cfg file for the MAC of the phone. I went into EPM and performed a "Save, Rebuild Configs and Update Phones" command, and it did update the date and time of the .cfg file. The files are there, but the phone is not pulling the file. I did a factory reset on the phone, put in the ip address of my server in the "Provisioning Server" entry in the Web interface, and rebooted. No luck. I am starting to run out of ideas.

Yealink T-32G won't provision!

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I have the Extension configured in FreePBX for PJSIP, which is on 5060, and the phone's setting for "Local SIP Port" is 5060, so that looks OK.

Module-Update from which URLs / IPs

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Hi all,

I want to secure my FreePBX-System by only allowing connections to and from known IPs (SIP-Provider, Telephones and systemrelevat IPs).

From which URLs (or better IPs) FreePBX gets its Updates?:
- Module-Update by Module-Admin
- System-Updates by System Admin Pro Module

Are there any other connections required back to Sangoma or FreePBX (e.g. for System Activation, Registration of Modules/Deployments).

Would appreciate any list of used URLs/IPs.

Thanks and best regards,
Patrick

Stay and away and app

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Hi Dave,
Maybe I was not clear enough, but my incoming calls are going to ringgroup .
And my thought was, that if one user uses follow me or don not disturb, the other user still receives the incoming calls !
Or am I wrong ?
Regards,
Harry

Bug or Design? Files erased automatically

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Yep, that's it - file names were 1_2345.wav for example
Version is 12


Bug or Design? Files erased automatically

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So is this fixable or should I keep my file out of voicemail folder?

WebRTC browser calls FreePBX 13 (Stable)

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My issue with JsSip returning 488 Not Acceptable was because it did not like the video offer coming from Asterisk. I disabled video and now can place and receive calls over WSS.

@tm1000 when you create the 99(extension) peer for the UCP WebRTC user, add "videosupport=no" to disable video in case the user has enabled it globally.

Bug or Design? Files erased automatically

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No fix for 12, you must:

  1. avoid underscores in filenames
  2. don't save in vm folder or subfolder
    or
  3. upgrade to 13

Follow Me Toggle Feature Code - Using from a different extension

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Hello,
If I'm at extension 123 and dial *21104, it toggles Follow Me for 123's extension and not 104's.

Is that possible at all, to toggle Follow me for a different extension than the one you are calling from?

dahdi_test timing _way_ off on vmware esxi 5.5, severe audio chop

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I could really use some suggestions. Conducting some tests today to see if how the VM behaves on different hosts; perhaps we can narrow something down.

We loaded the PBX on a new ESXi host; it is, in fact, the only VM running on that hardware. I've added similar high sensitivity options for this VM. I get better results, but still get things outside of range (less than 99.98%); some as low as 99.7%. I've gotten as low as 93% on my other VM, though (which sounds like garbage).

I installed a brand new, fresh PBX from distro on our problem host environment; right out of the box with nothing customized or running, we got audio chop.

Now is when I get really confused, because I did some tests in another environment, on another FreePBX distro VM, and the dahdi_test results were very similar (low of 99.6, cumulative avg of around 99.98). However, there is no chop in their audio whatsoever.

Did I mention this occurs on fully local connections? Two phones, 20 ft away from the pbx, making an internal call without our provider (sipstation) in the mix, and there will be audio chop. Not as much as a conference call, but chop none-the-less.

I'm really at the point where I'm about to give up and put the PBX on hardware. This has been going on for a month, and my customer is very unhappy. Even if I do so, I want to figure this out! Other people run their FreePBX on a VM (including me) with no issue. Some of those other people are my clients; I really don't want to start issuing hardware PBXs if this happens to my clients.

Are there any other causes for this kind of choppiness, outside of network packet loss? Is there anything I haven't tried to optimize this dahdi timing?

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