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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Video call software

In our company we have about 40 offices distributed all over the country. Every time the Human resources department want to do a training session the people have to go to the central offices for the ...
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1answer
165 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
157 views

How to add a user who is out of state as if they were just another extension AT our office? [closed]

We want to add a person who is in another state to our phone system as though they were simply in another cubicle here at the office. So, they can dial "302" and get the office assistant's extension ...
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2answers
62 views

How can I filter packets from a port monitor?

I have some data going from Point A to Point B. I have a SPAN monitor set up to a monitoring device C. To recreate some real world scenarios, I want to filter out all traffic which is a certain type ...
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3answers
358 views

How to monitor VOIP packets

We have Asterisk server configured in a Cent OS 5.5. Sometimes my colleague complains that the quality of the call is not good. How can I monitor the incoming or outgoing VOIP packets ? Is there a ...
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3answers
351 views

Bandwidth Suggestion

I have been asked to analyze the bandwidth usage of a company and make a recommendation for upgrading their Internet connection(s). Here is the layout 3 DSL lines so it is 3x(6 Down, 1 Up Each) into ...
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1answer
495 views

How to use skype as customer service tool

I would like to allow multiple support engineers to use the same skype account to communicate with customers. This also means multiple simultaneous voice calls with of the support skype account with ...
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1answer
605 views

VOIP phones stop working when spanning-tree (STP) enabled on switch

I have a few different types of managed switches, including mostly 3com Super Stacks, Baselines (for example, one specifically is a 3com baseline pwr-plus 2928). Most of them are configured with two ...
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1answer
451 views

Asterisk Connection not working

I have installed Asterisk on VirtualBox by following the steps from here. Everything went ok until I got to navigate to an IP to configure Asterisk using FreePBX: 10.0.2.15 (Shouldn't be something ...
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1answer
113 views

Turn old rack server into a telephone server

I have an old server lying around and I am thinking of using it as an internal telephone server. Its main use would be to set up a 1 to 1 telephone system that could be used internally to connect to ...
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1answer
212 views

GMAIL ports/protocols

I am trying to configure iptables on Ubuntu. I have the server setup as a transparent proxy using Squid which is working fine. At this time only port 80 is redirected to SQUID via nat iptables. I ...
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2answers
1k views

It is possible to set up an Asterisk PBX running on EC2 and have reliable call quality?

There is an EC2 kernel with a 1000Hz clock, which I have successfully used to set up Asterisk etc already, but I am wondering if other issues (e.g. transit from London to Dublin to London- our current ...
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2answers
201 views

VoIP and IPv6 with IPsec

I had some basic questions about VoIP in a IPv6 architecture, right now I'm running VoIP in a v4 architecture, and I was thinking about to change everything to v6. my questions are not that practical ...
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1answer
2k views

How can I link two Trixbox servers to make free calls between them?

We have a Trixbox server at the office, and I have one at my home too. They have been working fine independently of each other for a while, but I figured if they could talk to each other over the ...
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2answers
112 views

Recommended FXO Gateway for UK analogue lines

Can anybody recommend any FXO gateway devices to connect analogue telephone lines to an Asterisk VoIP system. Requirements: Minimum of 4 ports. Enterprise grade - quality is more important than ...
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2answers
6k views

What is the default username and password for the Cisco SPA-942?

I can't seem to find the default admin username and password for the web administration interface for the Linksys/Cisco SPA-942 VoIP phone. What is it?
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3answers
111 views

Is it possible to prevent certain PC's from using a specific router?

I'm setting up a PC & Phone network, and want to ensure that the PC's do not use the Phones internet connection. Is it possible to setup a Group policy statement ensuring they don't use the ...
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3answers
2k views

How do I setup a VLan? (Phone & PC Network, limited Ethernet Points)

how do I setup 2 VLan networks? What equipment do I need? I have a phone and computer network, phones need to be routed to the Phone server, and computers routed to the network server. The phone ...
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2answers
311 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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1answer
67 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
303 views

What is the best option for SIP service redundancy?

From your practical experience, what is the best option for SIP service redundancy? a) one SRV record pointing to a hostname which has two A records b) two SRV records pointing to two different ...
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2answers
349 views

Skype performance in IPSEC VPN

I've been challenged to "improve Skype performance" for calls within my organisation. Having read the Skype IT Administrators Guide I am wondering whether we might have a performance issue where the ...
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1answer
1k views

HP Procurve to Cisco - QOS

I have a HP Procurve with the following: vlan 4 name "Phones" untagged 20 ip address 10.50.60.1 255.255.252.0 qos dscp 101110 qos device-priority 10.50.100.10 dscp 101110 qos ...
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1answer
115 views

How to send Voice IP traffic over a different DSL router

We have the following equipment for our internet and voip access 2 x netgear ADSL routers 1 x DHCP / DNS server (Windows 2008 server) Numerous PCs VOIP Phones I want to configure our network so ...
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1answer
230 views

NAT setup for an SIP Registrar/Proxy

I am currently trying to get the following scenario to work. Warning: I am a software engineer--not a network admin. I have various SIP endpoints (sip based video servers) on a LAN. On a Windows ...
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1answer
508 views

IM and File Transfers Integrated With VoIP

I have an Asterisk box serving an office of people. I'd like to provide instant messaging and file sharing capabilities alongside the voice and video capabilities provided by Asterisk (ala Windows ...
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1answer
422 views

How to route VoIP phone through VPN

We have two offices (China and US) connected by site-to-site VPN using 2 Linksys RV082's. We also have a VoIP system with 2 VoIP phones in the China office. We have been having trouble with the ...
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5answers
612 views

How do I bond ADSL lines for VOIP?

We're using G711a as a codec, and I wanted to bond ADSL lines to give me the maximum possible upstream. I know that there can be issues of packet re-ordering etc; what is the 'right' way to do this? ...
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2answers
758 views

Sound disappeared on IVR in Asterisk

I'm setting up an IVR in asterisk. It was working fine but I installed some new hardware and the sound has disappeared. I can see this in the console: Feb 22 22:24:00 DEBUG[10555]: pbx.c:1697 ...
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1answer
697 views

Voip external extension behind nat?

I have a nice Talkswitch voip system. Per the Talkswitch specs, ports 5060 / UDP, 6000 - 6006 / UDP, 69 / UDP, 8484 / TCP, 9393 / TCP are all mapped to the Talkswitch unit. It has built in voip router ...
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2answers
824 views

open ports on a cisco/linksys wrt54g2 v1.5

Im upgradeing my router from a netgear fvs318 one of the problems im running into is on the old net gear i have opend ports under the "Add Service " for our voip udp 69 tcp 80 tcp 2000 udp ...
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1answer
866 views

Checking rtp stream audio quality

We are working in a test environment and need to monitor the audio quality of an rtp stream that is being captured using tshark. Right now we are able to capture the audio and access the file through ...
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2answers
739 views

QoS / PBR Routing Questions

I have a 50Mbs Satellite link and a 10Mbs Microwave link supplying a very remote location. Behind these links, I have a 6,400 seat network - with about 3,000 signed in at any one time. My goal is to ...
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3answers
116 views

Allowing phone network access to external mail

At work we currently have 2 separate networks, 1 for our phones (192.168.2.x) and 1 for the computers (192.168.1.x) that share 1 router (Draytek 2820) for internet access. We have a typical small ...
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4answers
2k views

Cisco router password change and now select VOIP phones not working

I manage this school network and I went in today to reset the password on their Cisco 2800 series router. I didn't have the original password on the system because it wasn't shared with me and the ...
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1answer
36 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
128 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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1answer
208 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 ...
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2answers
153 views

What determines the length of RTP packets?

I have established a SIP session between two clients. I observed the RTP trace between them on wireshark. The 'length' column has a value of 172 for the RTP packets that flow from Client 1 to Client 2 ...
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2answers
349 views

VoIP client permamently losing connection after ISP IP reset

I have a SIP client device connected to my ADSL router, so it is behind a symmetric NAT. My ISP resets the client's IP every 24 hours, it looks like this in the router log: Mar 26 12:36:56 daemon ...
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1answer
871 views

Poor audio quality during RDP with thin client from Windows Multipoint Server 2012

we have problems with audio quality from Windows Multipoint Server 2012 during RDP sessions, we are using Thin Clients for the Remote Desktop the display is good but the audio quality is bad, there ...
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1answer
64 views

Is there a way to get an “extension board” for a Polycom IP6000 phone? [closed]

I don't see any for sale, but is there a way to get one hooked up to it? Basically, I need 15+ speed dials. Something like this would be great, but for the Polycom instead of Grandstream: ...
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2answers
1k views

Remove local area code on outbound calls, and remove “dial 1” for long distance in Trixbox/Asterisk

We have Trixbox CE 2.8 and Asterisk 1.6. I want to remove the 933 area code so you can just dial 555-5555 instead of 933-555-5555 for local calls. I also want to remove the +1 for long distance so ...
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1answer
944 views

GUI for Asterisk 1.8.8 on Debian 6

Can someone point me towards a tutorial for installing either FreePBX, or a Webmin GUI for Asterisk 1.8.8 on Debian 6? Asterisk Thirdlane Multi Tenant PBX Manager and Asterisk Thirdlane PBX Manager ...
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2answers
283 views

Separate VoIP telephony and computer network - or not?

We have a PBX server (also connected to the internet) and all VoIP devices currently (early stages of deployment) set up on a network that is phsyically separate from the main computer network. As ...
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2answers
379 views

Asterisk and sip station codecs

we have a small instance running on ec2 of voxilla/FreePBX-Asterisk-1.6.0.5-i386.manifest which has been updated as far as i can see We signed up for sipstation (because it removes half the ...
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2answers
4k views

Asterisk - inbound calls from SIP DID trunk “rejected because extension not found”

I have a DID with DIDforSale that's pointed at my Asterisk server. When I call it from my landline phone I get the AT&T disconnected line recording. The Asterisk CLI shows the error message: [Oct ...
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2answers
1k views

Asterisk DID incoming rules =s

So I have a 10 way SIP trunk from a UK provider. Calls in and out work fine, but as I have multiple numbers for different purposes I set up the inbound rules for each number but they never work, only ...
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1answer
85 views

Linux audio capture at a rate of 20ms is choppy

I am currently working on an embed linux video server. When I deployed the application to the boards we are planning on using for production I noticed that the audio was noisy. I discovered that the ...
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2answers
245 views

IPv6 in a Small Business

I’ve been tasked to identify a new server and IT equipment for the company I work for (employees <20). I’m looking at getting a SBS 2011 Standard box. My Question is: Apart from ensuring that our ...