Questions tagged [voip]
Voice Over Internet Protocol is a name of a family of technologies that allows transmission of voice over an established network connection. What is usually discussed, though, is Internet Telephony - bidirectional voice (and lately video) transmission over public networks.
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Looking for a free VOIP solution using RJ11 PCIe card [closed]
After long hours and several days of diving layers deep through links from google searches, I've decided I just don't understand a lot of what is said about a lot of these "solutions". I don't know ...
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connect to voip gateway and wifi for internet on same windows machine
I have a situation which i will try to explain:
i have subscribed for a SIP line. In my country, it is delivered via a MUX device and a cat6 cable. I have connected that to my soft PBX which is ...
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How Does Making a VoIP call through PSTN Work? [closed]
for my college work i was told to draw a diagram showing "information flow" of someone making a VoIP call through PSTN and then explain how it works. After doing research for the last 3h hours stuck ...
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Understanding Yealink VOIP phone configuration
I know how to connect my VOIP phone to asterisk. My asterisk server sometimes goes down so I will like to understand better how can I use a failover server. This guide:
http://download.support....
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How do you fix "No Trust List Installed" error on a Cisco 7942G VoIP Phone?
I have a 7942G. I'm trying to connect it to Asterisk.
Looking at Settings -> Status -> Status Messages, I see this:
Error Updating Locale
SEP############.cnf.xml
No Trust List Installed
I'm only ...
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SOHO routers Simultanous VOIP connections / connection tracking tables and packets per second
I am planning on deploying VOIP for up to 6 simultaneous calls from VOIP handsets (through the Internet using a connection with a single world-facing static IP, over SIP, to a single IP address on the ...
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Dialer.exe in windows 10 | Alarm Call Out | Work with MS Access
Does anyone ever use and know how to configure dialer.exe in windows 10? I'm looking for an auto dialer that can integrate with MS Access to dial out certain users (at least able to add 3-4 users ...
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Asterisk call through browser
Currently, we are using two version of IPPBX (VOIP) System in our office, which are CUCM v11 and Briker 1.2.
We want to call directly through web browser instead of using a soft phone (or any desktop ...
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Internal calls issues asterisk vicidial
I have just installed new Vicidial v7.0.3 on an i7 12 GB Ram pc I have completed all steps in the installation including OS-install, network configuration, Firewall set to internal on one of the ...
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Route all traffic throuch a Linux machine
I want do route all traffic from and to two IP phones (SNOM D305) through a Linux machine for debugging purposes. The IP phones have a builtin 2-port switch each and the setup looks like this:
...
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How do I send Asterisk recording emails with a post-call script executed from FreePBX?
The script I have designated FreePBX to call, when called directly from the CLI sends the email flawlessly. The emails are not triggered after a call takes place, however.
Background context: https://...
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My voice comes out like a guitar on Discord
Discord (VoIP) never works for me. When I speak, people always tell me I sound like a guitar or a robot, or that my voice is really glitchy. I can hear them fine though. I have tried changing the ...
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Ekiga under Windows: registered but calls fail
I have a VoIP account with sub-accounts. Sub-accounts are assigned 3-digit extensions and I can call between sub-accounts using regular desktop phones connected to my ATA. Now I am trying to set up ...
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Possible to tracert Google Voice calls, if I route my phone# voIP on Linux? Linux Mint Rosa 17
Wondering if it's possible to tracert anonymous voip calls I'm receiving if I connect my phone number to a Linux voIP server (Asterisk, sipX, OpenPBX, etc.)? My idea is that I may receive info from ...
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Do I need to wire a CAT5e cable a certain way to provide digital TV over fiber optic?
Fiber optic internet, phone, and TV service comes in to an outside mounted ONT. Two CAT5E cables run into the home from 2 of the 6 ports on the the ONT, which I am assuming is one for internet and one ...
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twinkle configuration for Sangoma
I'm able to run AsteriskNOW virtualized on my local system. Also, I'm using Anveo on two different SIP hard phones. Anveo works fine, indicating that the network handles SIP fine.
Now, how do I ...
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WhatsApp low quality voice call on high speed LTE modem
I've this TD-LTE modem with internet connection speed of around 20Mb/s. With this connection, everything works fine except the WhatsApp voice calls. What could be the reason for delayed voice and ...
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How to setup a voip server connected to a ISP voice line
My home fibre bradband ISP provides me with a Huawei HG8240 optical network terminal. This has one ethernet port and one phone port. The phone port is active, but unused at the moment.
The ethernet ...
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Use of RTP Marker Bit
RFC 3551 indicates that it should be set only on the first packet for a talkspurt and that it must not be set if silence suppression isn't in use.
I am not clear though on whether or not there exists ...
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Asterisk (VOIP) port forwarding on TP-LINK
I have local Asterisk server on Ubuntu 16.04. Its address is 192.168.0.101:5060 and I would like to be allowed to talk with other users when I am outside local network. I tried this but it doesn't ...
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Bluetooth headphones connect to "Forward VoIP Calls to Handsfree" but not to "Listen to Music"
After restarting my computer, I can always connect my bluetooth headphones just fine. However, if they disconnect (intentionally or not) I open the device under "Devices and Printers" and connect ...
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What does the expected packet loss option in libopus (ffmpeg) actually do to the encoded file?
I am encoding some files into the opus format in ffmpeg using the VoIP application setting (this is for research).
One of the parameters you can set is Expected Packet Loss (%). I understand the ...
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Using a VoIP router behind a DrayTek 2760 router
I'm struggling with the following problem:
I use a DrayTek Vigor 2760 as router 1 to access internet by using a VDSL line. The DrayTek behaves as a Modem and DHCP server with IP 192.168.1.1. My ...
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I am having error in Asterisk
I am working with Asterisk 13 on OpenWRT(Linux distro) and having following error.
-- SIP/sipphone-00000000 is making progress passing it to DAHDI/1-1 [Aug 7 22:40:44] ERROR[13139][C-00000000]: ...
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How to whitelist IP’s on an Asus router to allow VOIP to pass through without dropped packets?
I recently installed my newly purchased an Asus RT-N56U and quickly noticed that I was getting disconnected from different VOIP services, namely both Mumble and TeamSpeak but otherwise my connection ...
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SIP based door entry and phone system
For a residential home, we've allowed for a door entry system, where we'd like to the following:
Push button at door entry panel calls all intercom stations (handsets) - audio only
From any intercom ...
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Asterisk ERROR: stasis_app_exec: Stasis app 'channel-dump' not registered
I'm testing Asterisk ARI and got this dialplan on extension 1000
exten => 1000,1,NoOp()
same => n,Answer()
same => n,Stasis(channel-dump)
same => n,Wait(100)
same => ...
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Why is the musical sound so much inferior to the voice sound on video chat apps?
My question is about the huge difference in the audio quality between the human voice and a musical instrument - for example, a piano - when using video chat apps as Skype, FaceTime, Facebook ...
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jitsi account "name" or "default identiy" sticks and cannot be removed
how do I change or remove where it shows a "name" above offline to the right of the silhouette of the user and below the files tools help menu?
I removed all accounts. I'm only using Jitsi for SIP ...
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PPPoE connection type with VoIP service
Two VoIP ATA adapters is used for VoIP service, adapters are in different offices, but within same VoIP service networks. The PPPoE connection type is used in ATA for VoIP service: is it possible to ...
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Connecting Teamspeak to Asterisk conference
I'm trying to connect a TeamSpeak 3 channel to an Asterisk conference (using ConfBridge) on FreePBX.
Not much out there about how to do it. Both the PBX and TS3 servers are running on the same CentOS ...
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Find out what winhook is causing mouse lag with voip software and 12 button mouse
Running windows 10, I recently got a 12 button mouse and I noticed that when i run any sort of voip software (Discord, Ventrilo, Skype, etc), my mouse clicks will sometimes not register. Or sometimes ...
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Unable to accept incoming calls on Polycom VVX 410
I'm using a Cisco DPC3848 cable modem running in bridge mode.
I have a microtik CRS125-24G-1S-2HnD router behind it.
I can make outbound calls on it no problem, but if I try to make an inbound call ...
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Why SIP-ALG in routers and modems?
We currently are using a NetSapiens based VoIP system and have extreme problems with a lot of consumer grade routers and modems due to SIP-ALG not being disabled or not being able disable.
So if SIP-...
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VOIP calls break up when browsing internet
I have been having a problem with VOIP calling on a number of different services including Skype, GoToMeeting, Webex, and XLite Softphone. The problem is this: normal internet browsing causes these ...
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Config to pass to Asterisk so it rings all extensions starting from "4" when a certain number is dialed
Which magic configuration line in which file should I put so that when a certain number is dialed, all extensions that match some wildcard ring, and a call is transferred to the phone which picks up ...
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Is Viber using TCP for VoIP?
I'm currently on a guest WiFi where all ports except those for surfing the web (80, 443, 8080) and mailing (25, 110, 143, 465, 587, 993, 995) are blocked (3128 seems to be working too).
I tried to ...
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What are the protocols that are used in teamviewer audio chat?
My country is blocking VOIP, so basically I can't make audio or video chats. However when using Teamviewer, it's always possible to make audio and video conferences.
I want to know how is that ...
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Linphone on Windows Server 2012, with multiple users
Im going to explain everything first. We have Windows server 2012, where our users are connecting to (lets say around 60-70 per server). After Linphohe installation and first start of the application (...
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Linphone SIP INVITE providing wrong IP on multi-NIC system
I'm trying to use Linphone on a Windows 7 machine that is connected to multiple networks (via 2 different NICs). It works fine with a single one connected, but when both are connected I only get one ...
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VOIP virtual numbers
I recently purchased a virtual number which is similiar to local landline number provided by my local provider.
how voip provider can provide virtual numbers over VOIP?
Do they have contract with ...
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Remove voice chat echo without needing headphones
When I voice chat with 3 people (using Discord/Teamspeak/Skype etc) and one of them talks their voice gets echoed from my end so the 3rd person can hear them twice. When I talk they can only hear me ...
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How to solve Win 10 "unable to establish networks connections"?
This has been happening since upgrade from Win 7 to Win 10. Service packs are up to date.
I use vBuzzer voip application as my phone.
Since the upgrade to Win 10, the phone does not work properly, ...
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Incomming local SIP calls. What's the cause?
I have Linphone installed on Linux. There is default identity set to [email protected].
I keep getting incomming calls from numbers like [email protected] or [email protected]. What is the cause of ...
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DD-WRT & Comcast SMC Modem
My office has a comcast business modem (SMC something) which is garbage and so I took a Netgear WNDR3400 and put DD-WRT on it to use it as the DHCP server. Per this article I added the router as a DMZ ...
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SIP ALG detector
I am trying to create a detector for SIP ALG. I really just need to know what the BEST method of testing this is so I can write a program to test whether SIP ALG is enabled or disabled.
Thanks
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Does Skype use only Ipv4 or also IPv6?
A simple question. Does Skype use IP version 6? If yes, how can i set it to use this type of addresses?
When IPv4 is not available, Skype seems to have problems:
Are there other VOIP application ...
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how do I provision a hard phone for getonsip?
In provisioning a Cisco SPA 942 hardphone, branded Linksys, how do I finish the setup for getonsip (or onsip)?
SIP Address: [email protected]
Username: foo
Domain: getonsip.com
SIP Password: ...
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SIP server name Defaults to $VERSION for Cisco SPA 942 hard phone
In reading the fine manual for a Cisco SPA 942 IP phone, branded as Linksys:
what is, in particular, meant by:
SIP Server Name
Server header used in responses to inbound responses.
Defaults to $...
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How to make international phone calls from my computer' through my SIP server?
I'm currently working for a scientific company, and I'm the only programmer here, and the company wants me to use one of their servers to setup a SIP server, and somehow be able to make calls to ...