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Voice Over IP

Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).  Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).

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Looking for a VoIP phone system.
This client currently has a PBX black box supporting over 50+ phones. They are renting 5 lines from a local teleco.
They are building a very modern 100K square foot facility and will be moving to there in the near future. I started researching a new phone system to replace the current one.
I am not a telephone guy, so I would like to hear your opinions. First of all, I guess VoIP is the way to go, correct? I am kind of an old school and did not trust VoIP at the beginning. But after using Skype calling long distance for years, I find it has better voice quality than the landline most of the time.
I heard of 3CX, a cloud based phone system with no need to run an in-house PBX hardware or software. If the PBX is virtually on the cloud, how can we connect the 5 rented lines from the teleco to the virtual PBX?    
What solution would you recommend based on your real world experience? Reliability, voice quality, easy management.

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I have a Cisco 2811 ISR that appears to only have 64MB in flash and just 33MB available.
I only need encryption K9 for SSH access to the box and I need to be able to send/receive
IP SLA checks for VOIP RTP. Can someone recommend which image I need to in download
for this? I tried to download Enterprise and that's nearly the full 64MB or more.
Need education on 5 WAN IP block (same subnet) and the MPOE running up a fiber connection to the office suite.    We walked into this situation illustrated below.  There is one circuit coming into the suite.   The internet service installed a 200 megabit fiber connection at the MPOE.  A couple businesses want their own separate public WAN IPs running off of this one circuit.   There is currently a couple TP Link routers that we like to replace.   What device (switch?  what kind of switch?  Any problems using one switch over another one?) do we use between the biscuit (one ethernet port) and the multiple WANs on the Sonicwall? Here's what we summed up the ultimate game plan below...

Use a Sonicwall Tz 500(a model with at least x8 interfaces) and configure 2 additional interfaces as WAN ports - this would then give us 3. Each of these we can configure with their own static IP accordingly. Next we would configure a LAN interface for each company. Then we would use Policy Based Routing to move traffic from example: LAN 1 "Company A" to WAN 2. Sonicwall also provides QoS I believe which will support VOIP traffic through the routing.

CUCM 10.5 SIP Trunking

    I have a two site Cisco Call Manager Phone System with one server at each site (FL and California).   I have had SIP trunking up and running in Florida for about a year.   We are in the process of migrating our PRI trunks in California to SIP trunks, but we are unable to complete the RTP (Voice) connections on the calls.   Every time we attempt a call the new sip trunk which is mapped through our CA Firewall, the Call Manager Server at that site advertises the RTP IP for the server in Florida.   Since these are not mapped through the other firewall, the call fails with no audio.   I cant seem to find a way to make the secondary call manager server advertise it's own IP address for the RTP instead of using the IP of the publisher.   The calls originate from the CA server, it is just the RTP that keeps requesting to send to the wrong server.   Any help on  how to force the subscriber to advertise it's own IP or how to change it would be greatly appreciated.   At wits end on this one.
Here is diagram for Voip phone connection.    phone1 ---- SW1 ----- Nexus7K ------SW2 ------ phone2
We configure auto Qos at two switches (SW1 and SW2), both switches could be 3560 or 4500 etc . Do you think we have to configure auto Qos or some Qos at the interface of Nexus7K which are connected to SW1 and SW2? Thank you
We have a Polycom VVX D601 IP phone AND we have a VVX D60 cordless phone that is supposed to register with the base phone.  It's not working.

We recently brought in some new fiber to the dry cleaners and with the fiber, our provider supplied IP phones (Polycom).  The dry cleaners need a cordless phone so the girls can walk around looking for clothes while talking to customers, so, we purchased a VVX D601 base and the VX D60 cordless phone that is supposed to work with the D601.

The cordless base will not pull an IP on the network.  We have plugged in the D60 to the router and it will not DHCP.  If we take the phone to our other office, an office with IP phones and a Xircom PBX, it pulls an IP.  If we take the D60 cordless to other networks that do not have phones, it will not pull an IP.  

Now, you're going to ask why we don't plug it into the switch with our other IP phones and the reason is our provider brought in a Juniper switch for the phones and they assign IPs on a static basis.  If we plug the D60 cordless into the Juniper switch provided by our phone provider, it of course will not pull and IP and our phone provider said they will not turn pan DHCP for us.  

So, my question is, since the base pulls an IP on network 1, which has IP phones and a PBX, but the cordless base will NOT pull an IP on any other network, is that because it's a phone and not a PC?

That's my guess.  The cordless D60 pulls an IP when plugged into a network with an IP phone PBX, but…
can I block all voip calls
I dont want to block one phone number at a time

I dont want calls from voip phone numbers because it is usually a scam
When booking a Skype enabled room that a Polycom Trio isnsigned in as, what are the expectations and controls over who can walk up to the phone and join the meeting?
It seems like a privacy/security issue.  
If you join the call from the conference room phone, how can you identify yourself as the organizer? Is the only way to do that to join from another device signed in as your organizer account??
I only want to use classic skype for windows 10

please dont recommend another product or another skype

very difficult to reach dial pad

when I am on hold, I am asked numerous times, do you want to keep on hold, press 1

and I need to click 5 places to see dial pad
Is there a shortcut
We are looking at moving from an in house phone system to a cloud based VOIP system. The issue is we only have one Ethernet jack at each workstation.  I have heard that it is not a good idea to connect the ip phone to the jack and then connect the workstation/laptop to the ip phone's Ethernet jack.  Does anyone have any experience with this?  Does it work or does it slow down the workstation, cause connectivity issues, etc.?  Any comments are appreciated.

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SolarWinds® IP Control Bundle (IPCB)

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We are adding a 4th switch to our network.  Not sure how best to configure.

Currently (see attached (new in red)) we have 3 connected switches:

HP 2910 L3 as core with two older v1910 and 3com 2952 switches connected via cat5.   The core switch routes traffic out to firewall.
The 2910 also has two vlans configured for data and voice.  The older switches are data only.

The new 4th switch will be in another office (c10-20m run away) and is a 2910al POE also.  We need to hook up poe phones here and desktops on vlans 20 and 1 respectively.

Main questions are:
  • Do we need two cable runs from the main 3 switches to the 4th?  i..e switch 1 to 4 and switch 3 to 4 (chain mode instead of looped?)
  • If two, can we mix the connections used to connect the switches i.e. fibre and cat5 or do they need to be the same throughout the switches e.g. cat5 only?
And if two connections do they have to be routing between the vlans e.g. vlan1 connected to vlan1 on switch 1 and 4 and vlan20 between switch 1 and 4?  Or does iprouting resolve this?

Hi All,

We currently have an issue with a new build at a remote site.

The overall voice network is fully working at other locations, however the new site is having issues with inbound calls from the PSTN. The phones at both ends (internal and external) will ring, however no audio is passed. The call remains open, but silent.

Calls work outbound from the site successfully. The CUCM/Cube are on the main site, where calls work fine. The remote site is connected to the main network over a site to site VPN.

The only difference between this and other sites is the allocated IP range. The Cisco phones on the remote site are all using public IP addresses, where the main network and other remote sites are utilising private address space.

Any thoughts or suggestions would be greatly recieved.

Many Thanks,

What is the difference between a Cisco phone "factory reset" and a "wipe" of the phone.

The code to begin the "factory reset" reset is: While the phone is powering up, and before the Speaker button flashes on and off, press and hold #. ...
Release # and press 123456789*0#

The code for the wipe is:3491672850*#

What is the difference?

Also, I have a Cisco 7960 phone that will not clear IP addresses and other settings for either process.

Any ideas how do completely clear the Cisco 7960?

Does anyone know what the state is with the Shoretel (mitel) director licences?  Particularly with extensions?  We need to add more and cannot find out much info about them.
Awaiting comms co to update me.
Same question regarding the phones.  I believe 212k are EoL.  We need the line keys so what options do we have?

NEC SL2100 phone system started behaving poorly a week ago Sunday.  Rebooted all network equipment, and nothing changed.  Reboot the phone system and everything worked for about 20 minutes.  Same cycle of events, about same time frame.  Next checked all network equipment for firmware upgrades.  Wireless bridge Engenius EnStationAC went from 2.x to on both sides successfully.  Problem returned yet again, but now a delay in getting dial tone of about 3-10 seconds.  If phone hangs up and connects immediately, no delay happens.  A monitoring computer is the only non-phone device on the entire voice network.  It shows random packet loss to phones.  The phone problem was originally they begain restarting at random.  They are not all happening at one time.  The delay never happens to VOIP phone on the system side of wireless bridge.  Engenius has yet to comment on the issue.  One phone (Only one phone) peforms IGMP actions accoring to Wire Shark.  Removing that specific phone solved the problem for almost 2 hours.  Yet it eventually returned.  All the time I have brought another laptop in for testing and moving it to both sides of the bridge.  If I am on the system side, I perform ping tests to the phones on the far side of bridge.  If I am testing the phone system resources, I am on the opposite side of bridge.  I perform around 1,500 packets to 3 devices simultaneously each time I test.  No latency or packet loss ever.  I have introduced a new PoE switch on the system …
Dear Experts, is this diagram correct?

We have 500 users, intend to use both VoIP service and Analog Tel service for backup. THere are 17 analog numbers to keep, so we choose Gateway GSM1024; for VoIP we use Grandstream UCM6510. We have several questions, can you please suggest?

- Can we use both UCM6510 and GSM1024 at the same time?
- Can we configure so that 1 department will use Analog line (GSM1024); other departments will use VoIP (UCM6510)?
- If one of these 2 fails , can the IP phone automatically change to the other, so that telephone service will NOT be interrupted?  
- Where can we find the reference link and installation manual for these devices?

Many thanks!
Wiping Cisco phones with the code: 3491672850*#.

How do I confirme that the phones are indeed wiped?
3cx.. moved VM from one host to another and set static MAC. still no ext and cannot create TCP connection to activation.3cx.com
Is the Jabra Pro 9450 Duo Stereo headset fully compatible with VOIP services like WebEx?
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Why Diversity in Tech Matters

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I have a Cisco 2960x switch and a heap of Nec DT800 handsets and would like to put them on a Voice VLAN. The only way I have been able to get it working was to enable the Voice VLAN on Switchport and configure VLAN tagging on the handset as we may need to use the built in switch.

Considering they move handsets often is it a safe assumption to just apply the Voice VLAN on all the switchports just in case they move one to an unconfigured port?

What’s the best practice?
I want a cheap 800 number service. 3 choices with prompts. All I want is voip. There is no call center. No forwarding to cell phones. Maybe just used for voicemail. A free 1 month trial and an expensive bill is expensive.
Hello and Good Morning Everyone,

          I recently added another phone line to my AT&T setup.  The hardware hookups are as follows:  The telephone/copier/fax machine has a direct telephone line connection to the back of the AT&T Gateway at the port labeled Phone Lines 1 & 2.  Then, I have another telephone line running from the Ext port of the  telephone/copier/fax machine  to the chordless phone.  That said, an AT&T agent explained to me that the  telephone/copier/fax machine with the direct connection to the Gateway will get the new activated number and the other phone, chordless one, will retain the old VoIP number.   Given this information, could someone explain what is meant by VoIP or Voice Over IP?  I did not want to bog down my AT&T agent with too many questions.  So, I decided to submit this one for review here.

            Any shared thoughts and explanations in simple terms with respect to the definition and function of VoIP will be greatly appreciated.

            Thank you

This is about the switch infrastructure using Cisco switches. Currently, there is only using one Cisco WS-2960x-48 POE switch. We also using Cisco UCS 500 series for the VOIP. We are using vlan 101 for data, and 102 for voice. Please see the attached cisco switch configuration.

Now, we intend to buy one new Cisco Meraki MS120-24 ports switch, and join this switch into the switch infrastructure. We also intend to add-in 2 more VLANs for our new VMware virtualization management and backup segments. This is a new 2-hosts virtualization (vmware), with 2 network ports to form a trunk carrying existing vlan 101 (data), management (vlan 121), and backup (vlan 122) from each host. How should I update in my existing POE switch and also the new Meraki switch? Can I make all the 3 vlans - 101, 121, and 122 routable but only allow selective ip to access. For example, only allow 192.x.x.25 to access all vlan 121 & 122 only, but not the other way round.

Thanks in advance.
Shoretel and switch STP on/off?


Looking at replacing our switches from procurve to Aruba.   Changing the method from daisy chained via ports to a stacked method using same models.  

Im unsure if we need to have STP disabled for shoretel to function?  If this is the case we cannot stack, which i find odd.

Hello everyone,
We have a ShoreTel VoIP phone system and we would like to use it to page different zones from the ShoreTel desk sets.
We have a 3 zone Valcom Page Control Unit.
Our paging goal is this:
  1. Page outside only
  2. Page inside only
  3. Page outside and inside at the same time
Is there a solution to accomplish these scenarios?
Do we replace our current Valcom 3 zone paging control unit with a new paging control unit capable of accomplishing the 3 scenarios?

Voice Over IP

Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).  Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).