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STRP with Yealink T4X phones

@sorvani wrote: Has anyone setup SRTP with the Yealink T4x series phones and FreePBX 13? SIP (PJSIP) over TLS is working perfectly, but I cannot seem to do anything with the RTP. Posts: 1...

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Fpbx 13 polycom vvx400 epm what is phone admin password

@ccts wrote: I'm provisioning Polycom vvx400's in the Commercial Endpoint Manager module & it's working great, the only thing not working is setting of the admin password on the phone. If I go to...

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Tranferring IVR from home built to Sangoma Freepbx 100 phone system

@Rednick1990 wrote: Hello All, I am getting ready to start up a new Sangoma FreePBX phone system 100. I already have a functioning home built FreePBX and I want to transfer the system over except for...

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Diversion Headers - Spectrum (TimeWarner)

@crbowers wrote: I was wondering if anybody has luck getting diversion headers working with Spectrum (formerly TimeWarner) SIP trunking? I did a search, and couldn't find anything definitive. I turned...

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Changing busy message depending on type of busy

@Patut1965 wrote: Hello, I am trying to make it so when a person enables DND the phone system doesnt say they are on the phone, instead it plays the person is unavailable message. The reason for this...

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Tone on attended transfer using yealink phones

@fmc wrote: Is there any way to play a beep tone when the receiving party is connected with the transfered party? Posts: 1 Participants: 1 Read full topic

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Incoming call transfer delay from an FXO to an extension

@citelum wrote: Incoming call transfer delay from an FXO to an extension Good Morning.My asterisk has an AEX410 card with 3 fxo + 1 fxs.The fxo ports are thus connected:1: direct to a pstn line2:...

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Freepbx version for trunk balancing module

@denismzs wrote: Hello I want to use the free module for trunk balancing. I have been using elastix 2.5 for a couple of years but I couldn't make it work so I decided to install a machine with freepbx...

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Error when entering new Sip Trunk :(

@duncanunderwood wrote: hi,I'm trying to enter a SIP trunk but every time i add i get this following error shown below.I think i need to delete a line in the database or something via the cli but I'm...

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How to upgrade apache?

@mvogel4949 wrote: This is probably a stupid question but is there an easy way to just update apache? Or if I run an update in System Admin will this take me to the latest version of Apache? My PCI...

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Help with Informational line setup

@striker495 wrote: I have created a virtual extension that has an IVR attached to it in the optional designations in advanced tab in extensions. When I dial the extension, it works fine. however when...

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Static call from yealink ip phone?

@the_passerby wrote: I have T28 sip phone and I want to call another sip phone without registration (I mean static call) how can I do it via web interface?Anyone can explain me? How to define callee...

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Random call drops and strange behaviour on Snom 300 handsets

@tigs77uk wrote: Hi FreePBX experts, We are experiencing some extremely strange behaviour with a FreePBX installation at one of our clients and are looking for some guidance on how to continue...

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Remove SIP Header

@Vargad wrote: Hi, we're having an issue when forwarding calls. Our DECT-phones (Mitel OMM) send a SIP header when forwarding calls ("Diversion", e.g.: Diversion: ;reason=unconditional )Our SIP...

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Using FreePBX 13 with a Mock Cellular Network

@Midimistro wrote: I am trying to set up a mock cellular network that simulates VoLTE by utilizing FreePBX as the SIP server. Here is my current setup: I am using an Agilent E6621A PXT Wireless...

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Cant dial an especific extension , app blackhole hang up

@Trezeguet17 wrote: Hi people i am trying to dial an extension inside of my pbx , but when i dial to it or some one frome outside call my number and dial the extension this one doesn't even rings , or...

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Could someone please explain why this follow-me won't work?

@maple wrote: I'm running Asterisk 13.16.0 & FreePBX 14.0.1rc1.17. This worked on an older version of FreePBX and Asterisk. Calling from extension 1100 to 1101. 1101 is on another call. Expected...

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Intermittent outbound calls fail busy signal

@sillsd4622 wrote: I've inherited a Asterisk/FreePBX phone system. I'm not a phone system guy but I'm trying to get familiarized. Last night we stated experiencing issues where some outbound calls...

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How to install freepbx gui on another ip and server is on another ip

@faisalkhan wrote: hey guys,how can i install freepbx gui on separate ip and server on another ip. just like the cloud environment like amazon ec2. the freepbx gui is on separate ip and server is...

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Res_Phoneprov provision

@dwsiemens wrote: Is there a way, or some module that lets you integrate the default provisioning with freepbx. So I currently have 3 asterisk servers (non freepbx) and I have about 400 polycom phones...

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