Questions tagged [voip]

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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Lync and conference phones - do all clients need phone numbers?

Bear with me here - very new when it comes to Lync :) We have a very simple installation up and running right now, with only Lync software clients installed. I've now been requested to install two ...
pauska's user avatar
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Using Asterisk and FreePBX how can I map extensions to outbound routes

I have a Trixbox server (Asterisk and FreePBX) that has multiple tenants on it. I need these tenants calls to go out via different outbound routes in order to split the bills at the SIP trunk provider ...
SimonJGreen's user avatar
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Adding more than one IP to sip.conf tcpbindaddr and udpbindaddr (Asterisk 1.8)

The defaults in the sample sip.conf under Asterisk1.8 contain udpbindaddr=0.0.0.0 and tcpbindaddr=0.0.0.0. I want to bind the incoming [foo] extension to udp:192.168.1.1/255.255.255.0, the outgoing [...
tarkaköpönyeges csecsfogdosó's user avatar
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Lync 2010 SIP Trunk Providers

I'm playing around with a small Lync implementation in my home office and would like to try replacing my 3cx server with it. Currently I use callcentric as my VoIP provider, but based on another ...
Joshua Dale's user avatar
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vSphere Promiscuous mode only receiving packets one way from network switch

We have two network switches, a POE switch (SwitchA) to power our phones / users computers and a non-POE switch (SwitchB for the rest network.) Each switch is setup to do port mirroring to support ...
steve.lippert's user avatar
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Asterisk IVR server in a Cisco VoIP system

Here's my situation: We have a Cisco VoIP system that we're very happy with, but I need to implement an IVR application on top of it. My thought is that a virtual machine running Asterisk and an AGI ...
failedassertion's user avatar
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How to setup multiple listening ip addresses on Trixbox

I currently have added another ISP to our Data Center and have decided to setup an additional ethernet alias on our Trixbox for the 2nd ip address of our new provider. From what I have read you can ...
Digital Sniper's user avatar
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Call issue with Freeswitch

I am testing the following with Freeswitch and different devices (nokia n900, nokia e60, ekiga) and have similar results between them. On the Freeswitch server (1.0.4 in multi-tenant mode) I have ...
gbraad's user avatar
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Moving from Analogue PBX to digital VoIP?

I don't even know if this belongs here?. If not, do let me know. So we have an analogue Alkatel PABX system in our little office. We have extensions, direct lines and PBX lines. We are trying to ...
0xdeadbeef's user avatar
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Skype companywide global contacts list

We are a medium sized company based across several sites and with a number of home workers. We have more or less settled on Skype as our defacto method of communication. At the moment the only pain ...
Martin's user avatar
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Can a OCS 2007 R2 mediation server be installed on a front end server?

Is it possible to install the mediation server component of Office Communication Server 2007 R2 on to the same hardware as the front end server component, or do I need additional hardware? If it is ...
Bryan's user avatar
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What are the most important skill/knowldege sets for a network administrator to have to support VOIP?

As a network administrator (Cisco switches, work with WAN providers, protocol analysis) I have an "area of responsibility" for the customer's VOIP project. The telecom group has primary ...
jpolache's user avatar
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Large Jitter Spikes on a Small Percentage of VoIP calls

On about 5% of our customer's calls, we see large jitter spikes and high delta #s that have caused an noticeable audible impact on call quality. (Stuttering/Breakups/Robotic Audio). We know this from ...
AVoIPm8's user avatar
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How can I tell asterisk to initialise each music on hold application when needed rather than constantly

I'm running a custom MoH application for a large number of users under an asterisk 11 box. The configuration is as follows: [customer-100] mode=custom application=/usr/bin/ruby /path/to/script.rb 100 ...
Andrew White's user avatar
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Linphone iOS behind CGNAT sets wrong Contact Header

We are using Linphone client on iOS over Verizon network. Our client sends wrong IP address on Contact header at 200 OK message as a response to INVITE message from Asterisk. SIP trace Our client's ...
Falieris Ilias's user avatar
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2 answers
370 views

Asterisk Transfer call to next extension if previous INUSE

I am trying to transfer call to next extension if previous is using (INUSE) or call is in progress. I tried to use EXTENSION_STATE(extension[@context]) to find the status as follow: [sales] exten =&...
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Cisco Voice VLAN with 802.1X Authentication

I've got a Cisco Catalyst 2960 that I'm trying to configure for a remote office with 802.1X wired authentication. The setup I'm going for is Switch -> VoIP Phone via Internal Switch -> PC/...
iamacarpet's user avatar
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177 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?
Steven McCann's user avatar
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3 answers
943 views

Asterisk and SIP trunking, is this configuration possible?

I am new to asterisk and before I dive in, I just want to make sure that what I plan to do is possible/correct. My office will run an asterisk server and have both local and remote extensions. We ...
Mike's user avatar
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Configuration of a Cisco IP Phone 7916 and 7975g

Hi everyone, I would appreciate some help configuring my Cisco IP Phone 7975. In fact the phone works well with the SIP firmware (version SIP75.8-4-1ES2) and an Asterisk Server. Now I would like to ...
Alex Strate's user avatar
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487 views

sip endpoint receives direct sip calls but doesn't receive redirects from the registrar

I have an SIP endpoint, which I wrote, that receives sip invites, processes them, replies, and sets up the rtp session just fine when contacted directly. i.e. sip:user@[actual ip address of ...
Jonathan Henson's user avatar
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How can I record a call transfer with freeSwitch?

I am using freeSwitch to terminate calls that originate on the local network from an IVR system. I have it working with several different voip terminators, and I can record sessions successfully, ...
Eric Z Beard's user avatar
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Configuring Hylafax to work with Asterisk over a single SIP provider

I'm setting up an Asterisk 1.8 based PBX, and as part of the requirements, I need to make provisions for sending and receiving faxes. The 'big picture' goal is installing a fax printer at each Windows ...
EB.'s user avatar
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Why do we even need a PBX software like Asterisk?

I am starting out with VOIP and configuring softphones. When we sign up to a SIP provider like FlowRoute we get the credentials for a SIP Proxy. Can't we just use the SIP registration details in a ...
Kramer786's user avatar
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6 answers
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How to record SIP traffic / calls for future auditing?

We have a VOIP (SIP?) phone system and have to record all calls for specific phones in the company. It may be required at a future date to listen to these calls for auditing. These phones happen to ...
Scott Szretter's user avatar
1 vote
2 answers
481 views

Advantages and disadvantages to using a VPS (or any remote server) to host a VOIP/PBX system

My company currently runs a Trixbox Pro server locally. We're getting sick of the licensing fees and terrible community support and are considering migrating to something like FreePBX. I see that ...
Eric's user avatar
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3 answers
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Corporate IM with video that actually works, suggestions? [closed]

Does anyone here have a suggestion for a cross-platform IM solution which will work with VoIP/video on both Windows (XP and 7) and Mac OS X from 10.4 and upwards? Right now, we're in a kind of mixed ...
pauska's user avatar
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Is there a stable cross-platform softphone? Or a Linux-only one?

We use an Asterisk based solution as our PBX, and although we've found hardphones work flawlessly, we have yet to find a reliable softphone. The basic requirement is that it allows multiple calls and ...
Ivan's user avatar
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Simple bandwidth monitor

I have a CentOS machine that is being used as a PBX. One interface is going out to the local network and the other interface is currently (pending equipment arrival) plugged directly into a Metro ...
yayim's user avatar
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1 answer
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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 ...
Sam Sanders's user avatar
1 vote
1 answer
120 views

I need some advice to come up with a network plan, or sort of anyway

At my workplace we have been allotted a few IP addresses by the ISP. I think currently one IP is accessible from the outside (by accessible I mean in a browser). Typing this IP into a browser gets you ...
zermy's user avatar
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2 answers
823 views

Is subnetting worthwhile\possible to seperate out VOIP in my situation?

I'm ballparking the numbers here, but I have 50+ machines and around 30 hard VoIP devices. Unfortunately, 10 or so PCs use the 100Mb pass-through port from the VoIP phones and 2 PCs use softphones (+...
Desvio's user avatar
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1 vote
3 answers
847 views

VOIP and internet connection speeds [cable vs. fiber]

Our office is migrating to IP telephony. We have less than 10 employees that will be using the phones. We currently have cable internet, and they just bumped the speeds: There is a data center ...
stormdrain's user avatar
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1 vote
3 answers
496 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 ...
Supratik's user avatar
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1 answer
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SIP [email protected] 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 [email protected]. What is the difference? How do they coexist/work together? For ...
Azendale's user avatar
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1 vote
3 answers
2k 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 => ...
Linux Intel's user avatar
1 vote
2 answers
5k 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 ...
chrism2671's user avatar
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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 ...
jpolache's user avatar
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1 vote
6 answers
737 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 ...
Doliveras's user avatar
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1 vote
1 answer
373 views

How to screengrab from Cisco SPA504g phone?

Is it possible to screengrab from a Cisco SPA504g phone? I'm wanting to make user manuals which have the exact thing the users will see. Would be convenient for remote checking of provisioning too, ...
jonathanjo's user avatar
1 vote
1 answer
1k 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 ...
CJ F's user avatar
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1 answer
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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 ...
Matt's user avatar
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2 answers
74 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 ...
engineerchuan's user avatar
1 vote
3 answers
381 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 ...
Campo's user avatar
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1 vote
1 answer
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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 ...
user avatar
1 vote
3 answers
617 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 wondering,...
Mohammad Alhashash's user avatar
1 vote
1 answer
5k 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 ...
Scott Szretter's user avatar
1 vote
1 answer
2k 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 ...
Tamas Ionut's user avatar
1 vote
1 answer
162 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 ...
Jake Andrew's user avatar
1 vote
1 answer
253 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 ...
Yves Richard's user avatar

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