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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SIP me@domain.com vs (123) 456-7890

I'm trying to understand SIP, and one thing that keeps confusing me is phone numbers and dialing plans vs SIP URI's like me@domain.com. What is the difference? How do they coexist/work together? For ...
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1k views

Elastix and extensions.conf

My VOIP provider required me to add this into /etc/asterisk/extensions.conf: exten => h,1,Hangup exten => _00X.,1, Answer exten => _00X.,n, Set(CALLERID(num)=96354205144239) ; exten => ...
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904 views

Dell equivalent of Cisco Span

We are in the process of upgrading out network to Gigabit to the desktop. Currently we are using this equipment... CUCM (Cisco Unified Communication Manger 8.0.3.20000-2) Cisco 2821 Integrated ...
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2k 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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1k views

Understanding VOIP “sniffer” traces

My customer is doing an Avaya VOIP pilot. I am looking at traces in Wireshark to build my troubleshooting skills. I have found much documentation on VOIP, but little that focuses on protocol ...
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722 views

Video call software [closed]

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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837 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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165 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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65 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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416 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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359 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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560 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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248 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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1k 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
735 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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136 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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219 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
220 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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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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124 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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10k 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
115 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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3k 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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1k 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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81 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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535 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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632 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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2k 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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158 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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941 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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1k views

3com 3101 or 3102 VOIP phones will not get DHCP/connect to network unless in specific location?

I am trying to figure out why at our company the 3com 3101 or 3102 phones we use, when NEW out of box will not jump on the network and update, unless they are at a specific physical location. For ...
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668 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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777 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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1k 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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846 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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863 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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2k 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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784 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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117 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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111 views

Asterisk with softphone and webphone

I have a problem with Asterisk. I wrote a WebPhone, so I should add to my peers some config details: encryption = yes avpf = yes icesupport = yes dtlsenable =yes dtlsverify = no dtlssetup = actpass ...
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95 views

VoIP calls dropping

Currently I have a proxy server running pfSense connected to a 40mb fibre lease line. This lease line has 60 external IP addresses (assigned to each room by a virtual IP in pfSense). On the LAN side ...
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1answer
287 views

why do all softphones fail to authenticate after receiving 401 unauthorized from Asterisk?

When I attempt to connect to the Asterisk server from CSipSimple, over a Cisco router, on the same network, Asterisk logs show the attempted connection: <-------------> [Mar 23 20:01:34] ...
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878 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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98 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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447 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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409 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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402 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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700 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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2k 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 ...