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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This is a general question to show my ignorant side.  We have construction going on at my business and they need to widen the driveway.  There are wires that need to be re-located for this construction project and the general contractor notes that there is 1 T1 lines and several fiber lines.  The fiber lines, I know, are run by the local cable company to provide us with digital phones and broadband internet.  I am guessing that the T1 is for the PRI for my ShoreTel phone system.  Is it possible to service my phone system some other way via those fiber lines or is the T1 the only way that I will be able to function?
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I'm receiving the attached error and would like to know how do you actually verify connectivity between these two? I mean the servers can ping and communicate on all ports, but is there a way from GUI/CLI to try to reconnect them?

We recently moved our CUCM 10.5 publisher to another data center. Call have been mostly good.
But we ran into a period where callers were getting this recording
"Call not allowed due to restrictions on your account". Can the Cisco
Unified Communications Manager 10.5 possibly be responsible for
that recording? Or would that indicate a problem at the provider?
I own an Avaya ACS509 R7 BUSINESS PHONE SYSTEM NAMED PARTNER. One of the extensions,the main programming extension number 10 is not functioning 100%.There is an illuminated button that can be designated as DND (Do Not Disturb). The button does make a click in the receiver but does not turn on the DND feature nor does the led come on. I have tried several identical phones on that  line and none of them work. All of them work different extensions. So, I must assume it is the main controller board. I am an Electronic Technician and probably can repair it IF I could locate a complete schematic diagram along with pcb board layout pictorials and part numbers/values. I do NOT need the programming manual or user's manual. ONLY a technical repair manual. They must exist somewhere since there are several locations that will repair this motherboard. I would really appreciate it if any Expert could direct me to a pdf or similar for this unit. It is also referred to as an ACS-R7 unit. Anyone assisting me will be the recipient of 735 virtual Kudos!

NOTE: This is NOT a voip system. Just a regular 20 extension phone system.
We have a UC520 ver 12.4

Currently I have a ephone hunt group that goes through a list of numbers to dial.
its tied to a external phone number door buzzer .
I would like to change that to a parallel so all extension listed ring but the parallel option is only available on voice hunt group not ephone hunt.

 ephone-hunt 1 sequential
 pilot ******** secondary 110
 list 103, 100, 125, 116, 109, 101, 111, 114, 122, 102, 127, 128, 107, 106, 105, 119, 108
 final 103
 preference 1 secondary 7
 timeout 20, 20, 20, 20, 20, 20, 20, 20, 20, 20, 20, 20, 20, 20, 20, 20, 20

here is an example of a internal hunt group (not tied to external number just extension 200)

voice hunt-group 1 parallel
 list 100,101,102,103,104,105,106,107,108,109,111,112,113,114,115,116,117,118,119,120,121,122,124,125,127,128
 timeout 6
 pilot 200

Basically I need to combined these 2 types of hunt groups into one that works for me.
Door buzzer dials phone number ******* which dials all listed extensions in parallel, and whoever answers first can buzz the person in.
I need a Google phone number that will ring on my cell number, for my business cards.

That gives me some flexibility if that number gets spammed, I guess.

Do I need Google Voice?

Could you provide me a link to get that number reserved? And was is Google Voice?

The topic is setting up remote admin GUI access for FreePBX on a virtual server. There is not an issue to resolve at the moment. The remote admin access will need to be set up in a couple of days and want to learn the correct config before attempting trial and error.
We have been using a different PBX app in the past, which was not compatible with a virtual machine environment. For that app, we had access to the admin interface via SSH port 443, and am not certain the same will apply for FreePBX
Thanks in advance ...
Is it possible to have a call attendant on Google voice , just even greeting or music?
Hello all,

I have some Win 2012 3cx v15 phone systems and was having trouble with apple push notifications for calls to remote devices.  I've determined it to be a TLS issue.  I had used IIS Crypto to remove the less secure SSL 3.0, TLS 1.0 and 1.1, leaving just TLS 1.2 and more secure ciphers.  This breaks apple push notifications from the 3cx server/software.  I put back TLS 1.1, no luck.  Put back TLS 1.0, now push notifications work.  I find it odd that I should still need 1.0 enabled on the server.  

Is apple push still using that protocol and not 1.1 or 1.2, or might there be something else going on here.

I'm by no means familiar with protocols/ciphers, just determined what fixes the problem.

we have two sites with skype for business deployed.

between the sites we have 2 subnets enabled via a tunnel.

we've discovered that although calls outbound via either site work calls between the sites don't connect.

looking at snooper logs we see that this makes sense by design as the primary site has a 'voip' subnet for the cx600 phones. when a call is made outwards between the sites and the user attempts to answer the call on the cx600 phones the call would fail.

when a call is made by either site outwards to the pstn calls flow as expected.

setting up the sites in the sfb admin portal i've setup the subnets which each site has.

based on my understanding media bypass should only work when the bypass site ID is the same. if it is not it should establish the call via the mediation gateway.

am i correct in this assumption?

i know i can add the non routable voip phone subnet to the tunnel however i'd rather not.
Free Tool: SSL Checker
Free Tool: SSL Checker

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One of a set of tools we are providing to everyone as a way of saying thank you for being a part of the community.

I am trying to setup our client side pc's to use the pass-through Ethernet port on our Avaya 1608-i phones. I am having an issue that I can only connect one device to the port on the Cisco switch at a time. If I plug the phone in to the switch it works and if I plug the PC in to the switch that will work fine. If I use the pass-though port on the phone then only either the phone or the pc will pick up an IP.

We are not using a separate Vlan for VoIP.

is there a setting on the switch that I am missing?

We have an On-prem shoretel system configured running the director version 18.xx.  We also have three shoretel switches and use both softphones and deskphones.

Shoretel Switches:

Ran through Communicator on PC's

Shoretel IP230

Edge Firewall:
Fortigate 100D

T1 Provider:

New WAN provider:
CenturyLink Fiber

# of Users:

Currently, our phones use a dedicated T1 connection through level3.  This T1 line connects directly to the SG-T1K.  Due to increasingly high costs, we are considering getting rid of the dedicated T1 line with level3 and routing our shoretel phones through our Primary WAN(century link fiber).  The fiber is connected to our Fortigate 100D edge router.  I have worked with shoretel for several years but I have not had to made a change like this before.  

My question is, can we accomplish this with our existing equipment(phones, switches, etc)?  If we can, how do i implement the changes.  If we can not, what needs to change or be upgraded to facilitate this change?

Thank you everyone in advance for any help that you may be able to offer.
I need to move a call manager which is a VM on a UCS C200. Can you please advise
on the proper shutdown procedure and the turn-up? It will retain IP address etc.
Just need to make sure I don't corrupt anything. I see a shutdown procedure
below for CIMC and using the power button. But should I also ssh to Call Manager
first and shut there as well? Thank you.

Several weeks ago I was surprised to get a Skype call from a relative who was traveling overseas. My surprise was not due to the fact that she was calling, but that Skype was running. I had not started Skype, nor have I ever configured Skype to start automatically at system startup! I thought it was just an anomaly, only to discover several weeks and at least as many reboots later that Skype was still indicating that I was online!!? When I check the usual places (the taskbar, etc.) I am not finding any indications that Skype is running. When I checked the services list, the only thing I am saying is the Skype updater. Does anyone have an idea what the heck is going on? I do not want Skype running unless I expressly permit it!!
Networking from the perspective of new phone system and QoS – not my area of expertise.   Yet, I have a customer that needs help and part of my job is to ensure that these changes do not impact production.  Which – has already happened.

Prior to my involvement, a 3rd party vendor made a change that caused a Spanning Tree event.  The vendor is Cisco certified.  Familiar with the “phone system”.

The AVPN is a managed network, the phone system is “managed”, and the customer dropped dollars for the third-party to make the necessary changes to the LAN equipment as well.  All Cisco.  Mostly 3750’s.

The network prior was not managed, earlier generation MPLS.  Upgraded to AVPN.  Some circuits higher CIR rates (my recommendations).  

The “old phone system” – working fine.  New phone system…. Static.  Calls dropped.  Voice mail complaints.  (short list).

There is more but I don’t want to reveal too much in the way of who or where.  This is not the first outage caused.  The managed circuit has FIVE 9’s and was down for almost 4 minutes last week – one outage.  No ticket generated.  No alert generated.  Customer of third party had to report it.  (Isn't 5 9's total of 5 minutes and 39 seconds for entire year)?  

This will get fixed.  

In building A, there are 10 floors.  Which they should already know.

One recommendation is to enable QoS on VOIP on the “trunk lines”.  Okay…… But why now? Why was this not done before when LAN changes were made.

I do not “yet” …
Hello all,
Here is the issue.
We have one user that uses a VoIP phone that is trying to go through our network.  He is a CxO and he is the only one that is using a VoIP phone.
He keeps saying that packets are dropping.
I just turned on 802.1p tagging but I do not know what else to do for a single phone.
Any help would be greatly appreciated.
Kelly W.
I recall a few years ago that I used a program similar to Skype which allowed me to have private conversations via this tool.

There was some sort of key that I generated on my PC and emailed to the other person, which that person added to this tool. We then have "private" conversations.

Does this sound familiar to anyone? That does tool, or another, offer this today?

Hey guys,

We have a FreePBX system, how can i access the recordings?
We are using Cisco Unified Call Manager.

Let's say John wants to call Jane. Both are corporate users but John wants to call Jane from his mobile phone to her mobile phone.

I understand Cisco have a plug in that integrates with Click-to-call API's and allows the creation of an app that performs a call back functionality the voice system will call back John, call Jane, and then bridge the two calls.

Does anyone know the name of the plug in?
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I learning up about Skype Enterprise Voice and it's potential for mobile.

Right now, we use Skype for Skype-to-skype calls. I'm wondering if this can be integrated with the telephone line in the office.  And then, if we can deploy a Skype mobile client to user's personal devices that has their office number....this way, we can :

1. A user never has to give out their personal mobile number, only their office number which will now ring on their personal phone

2. A user can make voice calls, either internally or externally, using this Skype client and reduce cost to them since it presumably uses the corporate network where possible, e.g. for international calls.

Some questions.

1. Is Enterprise Voice the term for the integration of Skype with the phone network

2. When we refer to the Phone Network, do we mean PSTN?

3. Is there a way for an enterprise Skype mobile client to have an external dialler feature so the user can phone anyone, either internal or external?

4. Are there potential cost benefits of this Skype client connecting to the corporate network

5. Is it possible for Skype mobile to have the user's office desk number so that it provides the fixed mobile convergence i talked about? Or does it need a separate number?
Plugging a Cisco 7941 phone in to an HP switch. Phone displays "Ethernet disconnected". Phone works fine when plugged into a Cisco switch. I've always had this issue with HP switches and just avoided plugging phones into them but this time its unavoidable. I've checked my Vlan settings and everything else I can think of. Does anyone have experience using HP switches with Cisco phone systems?
I have an analog door opener system in my building, this is the phone which opens the door: http://pasteboard.co/4JgctXi4x.jpg
I have a raspberri pi 3 model b, and a Pcf8591 ADC/DAC converter: http://articulo.mercadolibre.com.ar/MLA-641278286-modulo-pcf8591-conversor-4-ad-da-i2c-arduino-raspberry-_JM

What I want to do, is to connect the wires that goes to the analog door opener phone system into Raspberri PI, so I can hear/speak from a sip softphone installed in the raspberry pi.

Is this possible? I measured the voltage in the mic phone/speaker, it's shown in the picture. Not sure how to proceed with this.

Anyone? Thank you!!
MyNetFone is a VoIP company that allows an SMS to be sent by running the following:

https://www.mynetfone.com.au/send-sms?username=[username]&password=[password]&to=[destination]&subscriptionId=[subscription id]&text=[message]  

Open in new window

Details on this page https://www.mynetfone.com.au/Residential/Home-Phone/MyNetFone-VoIP/features/MyText-SMS/API

What would be the best way to trigger the above code when receiving an email with a specific subject or from a certain sender?
Thanks for your help
  • Have a deployment of 3 servers using dns loadbalancing.
  • we have a trunk setup via an sbc.
  • users have deskphones (CX600)

both inbound and outbound calls work as expected.

however, when a call is placed on hold after 30 seconds the call drops.

the same thing occurs when a call is parked..also after 30 seconds the call drops.

i should mention MOH IS SETUP in this deployment, however when an external call is parked there is just a beep sound.

when a call is parked between users MOH does work and the call does not get dropped.

refer and bypass are set to false on the trunk as well.  Trunk settings

in the snooper log i see references to "this call leg has been replaced"  in the same message as the BYE:
ms-diagnostics-public: 10026;reason="This call leg has been replaced";component="MediationServer"

the trace from the sbc shows that the mediation server is dropping the call so i haven't mentioned that here.

have the snooper trace if needed.

any suggestions appreciated.
How do I go about moving from a phone number hosted with Grasshopper.com to Google Voice?

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).