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*1 One Touch monitor need to disconnection only recording not call

@asifahmed009 wrote: Hi Guys, One touch monitor works fine.Means middle of conversations, activate the call recordings is working.Is it possible to end recording redial *1 or define any other feature...

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Outgoing registration status “Unregistered” after apply config

@cholegmw wrote: Hi,I have configured SIP trunk with registration.Everything was working fine till some other configuration changed and configuration apply (sip reload). Then outgoing registration got...

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Is it safe to remove these directories?

@dcitelecom wrote: I uninstalled the digium phones module, rest api, and rest apps and even though I selected remove from disk I have the 3 directories below on the disk. Is t safe to remove...

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Backup/Restore from v13 to v14 Failure

@RKM wrote: I have installed FreePBX 14.0.1alpha34. Prior to doing that, I used the Backup & Restore module, to backup my FreePBX 13.0.188.8 configuration. When I attempted to restore that backup...

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AWS PBX: can't register remote extension

@OLO wrote: installed freepbx on centos 7 of AWS, I can go to admin webpage via public ip, then add extension, however the extension can't connect server(public ip), also add security group for both...

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Sip_custom.conf don't work in FreePBX 13

@Driver wrote: Hello,On an older FreePBX version I was able to create custom extensions from the script and reload sip. Now the custom extensions are also visible in 'sip show peers', they are also...

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Backup / Restore fails with Permission denied

@dcitelecom wrote: First time I try to do a restore on another server of SQL table backed up with FreePBX distro and it fails with permission denied. Any idea what the error is? Starting...

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Error 255 Restapps Can't reload

@tigger1197 wrote: I just installed several commercial apps. Now when I apply conifg it fails and gives error 255 on restapps. exit: 255Unable to continue. SQLSTATE[HY000]: General error: 130...

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Jitter Buffer errors

@dcitelecom wrote: FreePBX 13.0.190.7 , Asterisk 13 I recently enabled the SIP jitter buffer and we are now getting the Warnings below with every call. Is this a bug or is something wrong with my...

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Playing MOH after a delay (timer)

@Issue wrote: Hi, Is it possible in FreePBX to Play Music on Hold (or announcements) after a delay, like if someone doesn't pick up the Phone after 20 seconds of ringing FreePBX starts an announcement...

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Compound Recordings

@mcisar wrote: I know there are several places within FreePBX where the system does not accept compounded recordings but I don't recall ever seeing the reason why posted anywhere. So now that I'm...

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"Ring Group & Page"... like "Park & Page"... feasible?

@mcisar wrote: Just wondering if any of the developers can comment on the feasibility of having a "Ring Group & Page" functionality, similar in concept to "Park & Page"; whereby an...

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Standard Opus bitrate on SNG7 beta release, Asterisk 13

@avayax wrote: What is the standard Opus codec bitrate on the new beta release of SNG7, Asterisk 13, and is it adjustable? By the way, I am trying to make Opus work on SNG7 and Yealink T23G phones...

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My ISP packages came with phone service, TV cable, and internet. Is there any...

@warheat1990 wrote: Hello, I subscribe to ISP that gave me internet, cable TV, and phone service. ISP gave me a fiber optic modem router combo which came with RJ45 port and RJ11 port. I plugged in my...

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Call to a busy DAHDI extension with no voicemail doesn't show up in CDR

@Marbled wrote: Hi! I noticed yesterday that a call to one of my DAHDI extension which doesn't have voicemail doesn't show up in CDR... Is this a misconfiguration on my side? by design? a bug? Any...

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Trixbox and secondary IP attached to main NIC

@friday1970 wrote: So, my fellow network admin and I have painted ourselves into a little corner. Years ago, we've set up a trixbox (v2.8.0.4), and for the most part, it still runs great. The phones...

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FreePBX 14.0.1beta1 UCP Loading Locks Page

@RKM wrote: I previously had FreePBX 13 installed and running UCP with no issues. In FreePBX 14.0.1beta1, when connecting to UCP, a "loading" modal shows and the web page then locks-up. This is in...

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How to extract caller id from P Asserted Identity for incoming call

@gooddisk92 wrote: Hi,I am able to capture the Caller ID in SIP Debug but I have no knowledge how to extract it.Therefore the pbx shows Anonymous from all incoming calls. The SIP provider require the...

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Vultr or Cyberlink?

@dcitelecom wrote: I know Cyberlink is the official FreePBX hosting provider so I understand that the official community needs to endorse them but I need a local IP in Asia and Vultr has servers in...

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Recording at /var/spool/asterisk/tmp

@batchen wrote: Hi, i have Asterisk 13.10.0 on SHMZ release 6.6 machine.we have a problem that too many recordings file types - wav , ulaw , sln48 ones in a while i see recordings are gatherd to 20G...

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