Voice over Internet Protocol (Voice over IP, VoIP) is one of a family of internet technologies, communication protocols, and transmission technologies for delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet.

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FreePBX UK Dial Patterns Route Through Specific Trunk

I've read a ton of links and tutorials and for some reason cannot get my dial patterns to work correctly. I've got Freepbx installed and working properly - no issue there. I want to create a dial ...
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2answers
443 views

SIP Trunk for VOIP behind a SonicWALL

I appreciate this question is quite "out-there" but - has anyone had any success with VOIP over SIP behind a SonicWALL. We are having an issue with one-way audio inbound. Essentially, the person ...
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1answer
1k views

Can't dial through SIP trunk: FreePBX/Asterisk

I set up a SIP TRUNK in FreePBX/Asterisk that works perfectly for incoming calls. Here' s the relevant configuration: type=friend host=201.217.31.10 callerid=mynumber ...
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1answer
41 views

Use VOIP through small switch

Probably a silly question but I have to ask. I don't know much about VOIP. We have a location with a large room now being used as an office for 2 people. There is one network cable ran to this office. ...
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1answer
350 views

Asterisk Sip Server and “Screen Sharing” function

I'am newbie in asterisk, i followed the tutorials on the asterisk wiki, and installed the latest version (13) of the asterisk server. I was able to setup the voip now i wonder, there is a function im ...
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126 views

Is it possible to virtualize a Genband/Broadsoft M6 voip telephony system?

I work with a company that runs a hosted VOIP platform originally known as the Genband M6 acquired by Broadsoft a few years back. This system is going end of life March 2016 and will no longer be ...
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29 views

How to automate updating of a setting on Cisco phones remotely?

Normally, to make a change one would get the IP address of a Cisco VOIP phone, access its settings via a web browser and make any changes to the settings they need to using the web interface. Is it ...
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1answer
709 views

Asterisk behind NAT sets wrong Contact Header

I'm using SIP with asterisk 13.1.0 behind a statically configured NAT. The servers private_ip differs from the public_ip, where I can reach it. I've already set these options in the sip.conf file. ...
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1answer
117 views

Ping time variation correlates with IP phones not working

We have an IP Phone system (most models are snom 320). Today someone's phone has decided to stop working. It has an IP address assigned to it by DHCP, and I can sometimes go to it's IP address to view ...
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2answers
191 views

routing for iax-protocol does not work on pfsense

I have problems with a pfsense-box routing iax-traffic (UDP port 4569). Our company has a central office and eight branch-offices. At the branch-offices we use pfsense as firewall and VPN endpoints ...
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2answers
768 views

Do I need VLAN for VOIP phone if my network switches supports IEEE P802.1p QoS?

I have 30 workstation and 30 IP Phones all under one single network 192.168.9.0. The network switches is unmanaged switch but supports IEEE 802.1p QoS (e.g.: DLink DGS 1008D and DGS 1016D). I ...
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1answer
1k views

Call duration limit & least cost routing

We have Elastix version 2.4 setup and we have some little questions which if you could please answer to solve via web based panel of Elastix, as agents are making calls via Elastix there are little ...
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1answer
307 views

Change vlan settings to Yealink phones at once

I have about 90 yealink ip phones. My target is to change vlan settings to all of them massively. For example, when it's going to reboot all of them, I run the following ...
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1answer
880 views

One-way audio problems with SIP over AT&T

I have a customer who has an VOIP PBX connected to a Level 3 fiber connection. He has offices all across the country using different ISPs. Two of those offices use AT&T, both in different states. ...
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59 views

Using a Barracuda F400 Firewall Router Without a Modem

I'm looking into a problem where VoIP phones don't ring for some of the incoming calls that I believe might be due to the network configuration. Do you need a modem if you are using a Barracuda F400 ...
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0answers
108 views

Cisco 7921G IP Phone not working

Got my hands on a second hand 7921G wireless IP phone. It should be configurable via USB; but need drivers to do that ...
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1answer
96 views

Configuring VOIP server without SIP

So I want to create a VOIP server with that I can connect with OpenVPN so I can call only a number of people that are in my network. I'll be using CSIPSIMPLE for that purpose. I'll be using only ...
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1answer
3k views

QoS settings for Wi-Fi calling on pfSense firewall/gateway?

The new version of iOS 8 as well as Android supports Wi-Fi Calling with multiple cellular providers including T-Mobile in the US. I am currently running pfSense as the primary Internet gateway for ...
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1answer
147 views

Preventing IP Phones from eating up DHCP addresses

We just installed a new IP phone system. The phones get their configuration via DHCP, which includes an option to set the phone to a different VLAN. The phone then requests an address via DHCP on that ...
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1answer
326 views

Asterisk relay between multiple subnets

I wonder what's the best way to go when you have phones on multiple networks which are not directly reachable. I have 3 networks 10.3.x.x 10.6.x.x 10.17.x.x My asterisk server resides on the ...
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1answer
174 views

How can I add user to Gnu Gatekeeper?

I've installed Gnu Gatekeeper from Debian wheezy repository. And I can't figure out how I can add user account to Gnu Gatekeeper server to use it in Ekiga or NetMeeting? What I should write in ...
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1answer
3k views

How configure 2 Lan cards in Windows 7/8 pc one to connect to Internet and other to Local Network [closed]

    I am about to install a dedicated VOIP server in our office. It is a 3CX pbx system on Windows 7/8 machine. The environment currently is a Windows SBS 2011 with 8 client machines. I want ...
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0answers
220 views

RTP analysis - Discerning ptime (packetization time) for a given VoIP packet capture

I would like some help on the subject of an automated way of discerning the average packetization time (ptime) of a VoIP call's packet capture. The reason I am not depending on the value in the SDP ...
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1answer
2k views

route incoming asterisk sip calls - fake auth rejected

we are trying to handle incoming sip_calls on our asterisk server, but somehow we always end up getting either 403 or 603 (which should be the same?) what should happen if it works: user calls ...
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1answer
403 views

qos with tc - high latency voip

I'm having following setup: local pbx doing calls for remote vpn clients. And have trouble with high latency a and quality of voip calls in remote office. Remote office is 10/10mbps link while local ...
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2answers
844 views

Asterisk inbound trunk unauthorised without allowguest=yes

I am trying to configure an Asterisk (Elastix) box to receive SIP calls from a provider without requiring allowguest=yes to be enabled in sip.conf. Basically the SIP trunk provider uses multiple IPs ...
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0answers
102 views

Switch from E1 to Analog PSTN line on IP-PBX failure

I have two Asterisk servers, E1(PRI) Gateway Asterisk 1.4 with E1(PRI) card IP-PBX Asterisk 1.8 on VM Server When my e1 gateway is failed (ex. shutdown, no power...), PSTN automatically switch to ...
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1answer
547 views

SIP INVITE packet has WAN address rather than call manager LAN IP

I am using SIP between two subnets (192.168.3.0/24 and 192.168.30.0/24) each connected via VPN. I have a call server on 192.168.3.100, and two phones 192.168.30.118 (Ext. 3128) and 192.168.30.119 ...
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1answer
706 views

Freeswitch : Audio handshake failure 1 error while making call using Sipml5

I just installed FreeSwitch and successfully connected to server with user 1001. Details -> OS - Ubuntu 12.04 LTS 64 bits FS - 1.5.13b+git~20140614T114905Z~fc7a74905b~64bit OpenSSL - 1.0.1 chrome - ...
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1answer
2k views

Voice VLAN Configuration on Cisco SG300

We currently have a single Cisco SG300 switch on our data network. We are introducing a hosted VoIP system. As such we will be installing a second Cisco SG300 (to account for growth and new phones) ...
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163 views

Cisco VoIP telephony - Can't save Night Service Bell Configuration

When trying to save "Night Service Bell Configuration" settings under Cisco Unified Communications Manager Express the following error message pops up. Configuration change failed. Please report the ...
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1answer
716 views

QoS Degrading VOIP

I have the privilege of trying to optimize our network at work to help improve the quality of the voip system we have. The general setup is: Bandwidth is 18 Mbps down, 3 Mbps up. ~10 people in ...
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1answer
125 views

Is running our workstation network through VoIP handset best practice? [closed]

We are purchasing a Cisco UC VoIP phone system and the handsets come in two flavors: 100mb and 1gb. Both allow network pass-through of another device such as a printer or workstation. Is this an ...
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0answers
208 views

Werbrtc2sip crashing intermittently. Aborted (core dumped)

I am having issues with making webrtc2sip actually work. Webrtc2sip is now getting crashed intermittently during a call. I have seen this even when only one call is active in a queue. By the way, my ...
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1answer
99 views

Normal phone number over ethernet, what is this service called? [closed]

Plug in phone to wall with ethernet cable Someone calls a toll free number Your phone rings What is the correct, vendor-neutral term for this service? Some terms out there are: Comcast calls ...
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1answer
2k views

DHCP For Voice And Data VLANs With Router-on-a-Stick

Background: I have 2 Powerconnect 62xx switches stacked with a Watchguard XTM-26 between them and the outside world. Currently all devices connected to the switch are in the same vlan and the ...
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0answers
394 views

freeswitch forbidden gateway error

i've stuck into a strange problem, i have Ubuntu 12.04 with static IP, installed Freeswitch with FusionPBX with easy install script from here http://wiki.fusionpbx.com/index.php?title=Easy_FusionPBX ...
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232 views

Connect Cisco 7975 VOIP phone to two separate Asterisk PBX's

I have a Cisco 7975 registered on an Asterisk SIP server, and I want to register a second line on a second (different IP) Asterisk SIP server. I tried to do this with firmware version 8.5.3 but it ...
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3answers
1k views

Connecting VoIP Phones to a Laptop

I'm trying to connect two VoIP phones to a laptop using the internal ethernet adapter and a usb-to-ethernet adapter. My configurations are as follows: Laptop eth0 - 121.1.2.1/24 Gateway - 121.1.2.4 ...
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797 views

Soft phone not getting register to my asterisk server

I have config in my local lan asterisk for test. which is working fine. With the same OS version and packages on "AWS Cloud" I am setting as following. But it don't get register "error = sip 408 - ...
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1answer
471 views

Security, hardening and NAT for RTP and SIP

I decided to create a VOIP server and it has gotten more complicated then I thought it would. So in order to make my server more secure, I keep it behind a pfSense appliance. I use IP filtering to ...
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1answer
11k views

Cisco ASA 5505 - NAT or Port Forward for SIP / VoIP ver 8.4

I just had an NEC PBX installed that lets me use SIP trunks for VoIP services, My gateway is a Cisco ASA 5505 running 8.4 and I only have one public/static IP Addresses. So far, my trunks are ...
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3answers
2k views

How do I calculate the necessary bandwidth for my office with PCs and VoIP phones?

In our office we have 30 PCs, 2 WiFi and 6 VoIP phones connected to the network that consumes bandwidth. Basically 30 PCs are simultaneously using the Internet (like Facebook, surfing the web and ...
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3answers
230 views

What is the use of echo canceler in ISDN boards if all my endpoints have echo canceling?

Most digital telephony boards (BRI/PRI) have optional hardware echo canceler presented as alternative to the cpu intensive software echo canceling available in Asterisk and FreeSWITCH. I'm ...
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2answers
167 views

Asterisk, GSM-FXO gateway example of usage

Can anyone explain where to use GSM-FXO gateway. I understand where to use GSM-FXS - for example to connect analogue telephone to gsm gateway. But do we use GSM-FXO to connect to two telephone ...
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91 views

How to set Mitel 5224 ringer volume in configuration?

We have several Mitel 5224 SIP phones, but can't seem to find out to make the ringer volume setting persist after it reboots. Can the ringer volume can be set somewhere in the XML configuration file?
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2answers
940 views

Setting up VOIP system on separate VLAN / Server on default VLAN

First off, I am not a network engineer so apologies if I get terminology messed up, i realize accuracy is important here. Currently, we have a small(ish) network with around 40 devices and 20 IP ...
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717 views

Cisco SPA504G phone CDP/LLDP issue

Good morning everyone, Just recently signed up for the site because I'm looking for some help with an issue that's been plaguing me for about a week now. I hope I'm posting in the right place. If ...
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1answer
80 views

Which one VoIP SIP audio codec should I choose to high quality calls?

The quality of VoIP calls depends on the codec used for the transmission and on the bandwidth of Internet connection. I want to use best optimized codec for my Internet connection. I have to select ...
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1answer
384 views

Asterisk ignoring sip_domain part of URI

I am using Asterisk version 11.5. My problem is the following: I literally cannot make any outbound call. That's because, whatever I dial in any client, the sip_domain part gets ignored, it is ...