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Call drops on attended transfer

@nickb wrote: I’m having an issue with calls dropping during an attended transfer with an external call. Internal calls transfer normally. The system is a fresh install (yesterday) of FreePBX 14.0.5.2...

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Cannot call to external ringgroup

@Dutch_voip wrote: Hello all, I have 2 freepbx boxes, one local and one remote that connect to each other using a siptrunk, I have a few questions on these… i need to be able to call ringgroups on the...

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Huge list of warnings in the "log files"

@Tommaso wrote: Hello everyone! another day, another noob question! Today i’m here to ask if someone is so kind to read my logfiles and tell me if i have to worry or not, because there is a HUGE list...

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TFTPboot directory missung files?

@bpatterson117 wrote: I am working on provisioning a bunch of grand stream phones and was doing some troubleshooting. I enabled Verbose logging on the TFTP server and was watching the requests come...

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The outdoor call does not work

@Cricchetto wrote: Hello everyone, suddenly the call received from the outside no longer works. After 2-4 seconds, hang up. I saw in the CDR report that passage is the number but I understand little....

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Erorr with Freepbx gui (Non-Commercial)

@irinetsys wrote: Just want to say upfront thank you for anyone that is able to provided guidance with this issue that I am having with my freepbx system. I have having an issue where I can not at all...

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End Point Manager, timeline for Digium D80 support?

@kombi1 wrote: Is there a any news as to when Digium’s lovely D80 handsets will be supported by End Point Manager? We have a number of those and would happily buy the commercial license. Cheers,...

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Inbound route call playback

@akam wrote: Hello! Is it possible in freepbx 14 to give access to someone in ucp panel to list and playback recordings to whole inbound route? Posts: 3 Participants: 2 Read full topic

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AOR Not Found?

@bpatterson117 wrote: When I boot up a new handset that has a extension mapped to (in this case 3201)it the asterisk log says [2018-12-11 07:19:39] WARNING[10194] res_pjsip_registrar.c: AOR ‘3201’ not...

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Stopping beep and announcement in chanspy

@dgilvin wrote: I need some help in getting rid of the beep and the message telling me what ext. when listening in on an ext. I am using Freepbx 14.05.2 Asterisk 13.22.0. I don’t want a beep to be...

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Alternate extension for voicemail?

@snaplink wrote: I have an employee leaving the company and it’s been requested that his extension stay alive for 30 days in case he receives a call. It’s also been requested that if the caller leaves...

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FreePBX 13 IVR direct dial extension to other pbx

@cnissarte wrote: Hello, i am a beginner with FreePBX or any IPBX based Asterisk Presently, we try FreePBX in our organization, i install and configure this system, configure an sip integration with...

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Freepbx machine based on raspbx ... FreePBX Framework error [SOLVED]

@claloano wrote: Freepbx machine based on raspbx … FreePBX Framework error Version: Asterisk 13.20.0 FreePBX 14.0.2.10 raspbx-04-04-2018.zip And load “raspbx-upgrade” for update He asks me to...

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Time to answer a call recording?

@sxarthur wrote: can i edit into CDR or any other module how long it takes takes to a person on an extension X answer a call? Posts: 4 Participants: 2 Read full topic

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Hostnames in Intrusion Detection FPBX V14

@jtuttle wrote: Hello, are hostnames (e.g. location1.domain.com) and IP addresses (64.25.14.1) both allowed in the Intrusion Detection whitelist? Need to always permit the premise phones to access the...

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Route based on current queeue hold time

@dannyprecise wrote: I see we can route to full destination based on X amount of callers in the queue. Can we do that if caller A tries to go to Queue 123 but and route them elsewhere if the longest...

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FreePBX as Outgoing PBX for another (old) Asterisk

@betamax wrote: Hello, i have to switch from Asterisk to FreePBX in two Steps (i also change the Provider) Old Asterisk = 1.4.21 FreePBX = 14.0.5.2 What is working: The Trunk from FreePBX to the new...

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The number you have dialed is not in service

@calida82 wrote: Hi I installed free pbx on raspberry and I’m trying to configure the cenbtralino. I set 2 extensions on two phones with zoiper and I can make calls between them. with zoiper I can...

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Remote extension RTP traffic being sent back to wrong IP

@bpatterson117 wrote: Hi All, I am using an external extension and I am running into an issue where my RTP traffic for a SIP call is being sent back to an unknown IP address. I have Nat configured...

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Music on hold only for users setup

@Cricchetto wrote: Good morning, I can not figure out how to set the music to wait for the conference. I set in conference “music on hold” on yes and “music on old class” on default. In this way, the...

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