Text to speech using AMI?
Reddit » Asterisk
by /u/xIilfly8462412989
5M ago
I've been reading a couple different methods for generating text to speech, but still kinda confused. It looks like Asterisk can install a text to speech engine. Is there a way to generate an AMI command that saves the message as a string in a variable and initiate a call to an extension that then plays the voiced sentence from the text variable? https://preview.redd.it/h9ak09wi0qrb1.png?width=1632&format=png&auto=webp&s=48036c90cd9146e3fc4f1c17ca1657391e332471 ​ ​ ​ submitted by /u/xIilfly8462412989 [visit reddit] [comments ..read more
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How to make phone Polycom phone auto answer?
Reddit » Asterisk
by /u/xIilfly8462412989
5M ago
I have an older Polycom IP330 phone and FreePBX. In PBX for that extension, in the advance tab, I have the "Internal Auto Answer" set to Intercom. But calling that phone from another internal extension still rings, it doesn't answer answer to intercom. I'm trying to not have it ring at all and go right to intercom. Is there some trick to it? submitted by /u/xIilfly8462412989 [visit reddit] [comments ..read more
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When a call is made, hang up right away - and capture the event?
Reddit » Asterisk
by /u/androidusr
5M ago
I want to ring a bell when a SIP call is placed from one particular client. So the process goes a little like this: When a SIP client calls, hang up on them right away. Make either an HTTP API request, or publish something to MQTT server ESP8266 (which can subscribe to the MQTT topic) rings a bell. On the Asterisk side, can you give me some terms to google so I can go about figuring out how to hang up on the call and also make an API call or MQTT publish? I'm using Incredible PBX, and I'm still a newbie. Thanks! submitted by /u/androidusr [visit reddit] [comments ..read more
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Asterisk for baby audio monitoring - FreePBX or some other way?
Reddit » Asterisk
by /u/androidusr
5M ago
I'm interested in setting up Asterisk server to use a bunch of voip phones as baby monitors. I'd like to be able to use the buttons on the voip phones themselves to start routing the audio from the baby room voip phone to say the kitchen or the livingroom or basement. My homelab equipment consists of a PC running Proxmox and a couple of rasbperry pi. I was wondering how best to go about installing Asterisk. Since I've not gone through the process of configuring things, I don't know if the other things packaged with FreePBX would be crucial to what I'm trying to do? I would love to make use of ..read more
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AMD detection with Stasis
Reddit » Asterisk
by /u/itsmee15
5M ago
Hey everyone! I am facing issue with AMD detection with stasis, as far as I have researched there is no way to detect AMD with stasis application, Basically I am doing everything like originating call and adding channels on bridge and 2 way communication through stasis, using Twilio as SIP trunk provider. Now what I am trying to do is moving from stasis application to dialplan for AMD detection and then moving back to stasis from dialplan with appropriate AMDSTATUS, and then in stasis they both will be handled accordingly. But the issue is when I move from stasis to dialplan I see this: -- Ch ..read more
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Phones becoming instantly unavailable
Reddit » Asterisk
by /u/Ok_Improvement7107
5M ago
Running IncrediblePBX on a VPS in Germany. All phones are outside of the internal network with their IPs whitelisted on the server. Devices register just fine, but upon calling them, the IVR says the person is unavailable and goes to voicemail. Log file shows this [2023-09-21 04:48:40] NOTICE[2800] chan_sip.c: Peer '702' is now Reachable. (125ms / 2000ms) 4[2023-09-21 04:49:44] NOTICE[2800] chan_sip.c: Peer '702' is now UNREACHABLE! Last qualify: 125 Any help would be greatly appreciated! ​ submitted by /u/Ok_Improvement7107 [visit reddit] [comments ..read more
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Asterisk 16 (freepbx15) and AWS
Reddit » Asterisk
by /u/Jotha-7Rubio
5M ago
Is there someone with FreePBX 15 in AWS? I have some audio issue that I can not fix yet. Nothing about RAM or Disk or the SIP Trunk. Is there any issue with AWS and FreePBX??? submitted by /u/Jotha-7Rubio [visit reddit] [comments ..read more
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Anyone using Twilio for trunk over tls with pjsip?
Reddit » Asterisk
by /u/CryptoFarmer1776
5M ago
I have found many guides on encrypting tunnel endpoints, but they are mostly all for chan_sip. Since Twilio support doesn't seem to have an answer, and following their guide did not work - just curious if they even support pjsip over tls transports? submitted by /u/CryptoFarmer1776 [visit reddit] [comments ..read more
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A multi tenant solution
Reddit » Asterisk
by /u/ahmadafef
5M ago
Hello, I'm looking into providing VoIP service to some of my clients. This i what it should be: 1- I have multiple clients. 2- Each client should have a different phone number 3- Each client might have internal extensions 4- I must have a way to limit the outbound minutes each one of them can use. 5- Each client should have the ability to set IVR and waiting tones. I prefer a reasonably priced safe and secure solution that is also easy to deploy and use. I appreciate your help! submitted by /u/ahmadafef [visit reddit] [comments ..read more
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Ata voip to analog converter
Reddit » Asterisk
by /u/Stock_Individual_245
5M ago
Looking for good ata i know grandstream make good ata but looking for any other choices that are better then grandstream ata with 8 port fxs port.had some issues with grandstream with couple of ata so looking for new company.any recommendation will be appropriate. submitted by /u/Stock_Individual_245 [visit reddit] [comments ..read more
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