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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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What kind of method should I follow to make 2 people to talk on network? [on hold]

I am developing a client - server application which client is android and server is a pc. I need to make them communicate with their microphones. What kind of method should i search for ? VOIP ? ...
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Correlating multiplex RTP streams to SIP SDP in VoIP

I'm trying to understand more about RTP multiplexing, i.e. where a single RTP session defined by its 5-tuple is used to send more than 1 RTP media streams. I've read that this could happen in cases ...
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No response when access web administration page of wireless router using popular browser, is it a phenomenon of hardware failure?

We have an entry level enterprise wireless router TP-Link TL-WVR1300G (Chinese page) in city B, and we have two physical servers located in city A and city B, and we setup a VoIP (SIP) trunk through ...
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38 views

Asterisk sip.conf in MySQL database

I'm able to include a phone inside a database for realtime usage. So, this code (from /etc/asterisk/sip.conf): [phone] type=friend username=phone secret=12345 host=dynamic disallow=all allow=g729 ...
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1answer
92 views

Periodic Internet Disconnection

Here is my problem: A) Goal Fix periodically disconnecting internet connection. I.e. maintain constant connection to internet (browsing etc.) while wired to router (ethernet). B) Problem I've ...
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56 views

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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11 views

How to connect an IP Phone to a different network.

So I have this problem , I have a really enthusiastic client who wishes to setup an IP phone to our netwowrk. I know it’s not that easy as it seems. He has his CUCM server in the US and want to setup ...
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41 views

Is there a way to get Avaya to stop lowering the volume all the way on every new call?

Everytime we make a call with the Avaya client here the volume gets lowered in the Windows 7 mixer for just the Avaya mixer. Is there a way to keep the volume up in the settings? Someone told me you ...
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25 views

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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35 views

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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482 views

How to find out what's preventing the Steam's new VoIP from connecting

As you might already know Steam has updated their whole chat design and system, but there is a problem with it. I can't connect to it, whenever I try to connect it stays at "connecting..." forever. ...
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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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411 views

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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49 views

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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637 views

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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344 views

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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48 views

Radvision VC240 - SSH and HTTP access

I've searched all the superficial web looking for the default login and password for the Radvision VC240, a videoconference terminal, for access via a SSH client. Need to find out if the VC240 stores ...
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1answer
166 views

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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32 views

Can I fool (spoof?) a pre-provisioned ATA with iptables or a hosts file?

I have a SPA122 ATA that is pre-provisioned to only work with "company A", a VoIP company that I was formerly getting service from. The web interface shows the non-editable setting "Restricted Access ...
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21 views

Does hanging up Skype call and restarting actually improves connection?

Sometimes we start the call and connection is bad, but then we hang up and call again and somehow call quality is better. Why should this matter? This might be relevant to any VoIP. Doesn't Skype ...
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129 views

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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373 views

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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1answer
70 views

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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39 views

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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212 views

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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1answer
280 views

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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380 views

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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255 views

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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1answer
198 views

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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1answer
418 views

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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573 views

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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1answer
417 views

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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1answer
101 views

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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243 views

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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728 views

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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1answer
408 views

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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2answers
89 views

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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1answer
39 views

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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1answer
310 views

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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1answer
250 views

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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130 views

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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170 views

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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1answer
616 views

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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2answers
478 views

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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1answer
132 views

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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1answer
4k views

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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257 views

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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171 views

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 (...