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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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I do not have an account yet. Where do I sign up?

I also have an iPhone 7. Once I register, do I download the app and just log in?

I need to share my desktop for a presentation, so hopefully, WhatsApp supports screen sharing, like Skype. I have a Mac and want to present a Powerpoint Presentation running in Windows 7 on Parallels.

For this to work, can I install WhatsApp on my Mac? Then, load Parallels, Windows and Powerpoint?

Or, do I need to install WhatsApp on Windows?

Thsnks
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So we have a client who would like to have different ringtones for internal and external calls, because when a call comes in, all phones ring, and they generally ignore it unless they recognise the number and then someone at reception picks it up and transfers the call to the right person, and they just think its another external call and ignore it, they cant tell the call has been specifically transfered to them.

The system is Elastix 2.3 and the phones are Grandstream GXP2130.

I found in the GXP2130 settings I can set a different ringtone for a certain number, so I can type "XXX" as the matching rule and all 3 digit numbers (All extensions are 3 digits) will then ring with a different ringtone. That is working fine.

But the problem is when reception picks up an external call and then transfers it, the caller ID is forwarded when the call is transfered and the phone then see's the full 10 digit number as the ID and rings with the normal tone.

How can I stop caller ID from being passed on when a call is transfered?
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I'm running CUCM 9 and Unity connection 9.  All screen's on my Cisco IP phones go dim (black) at 5pm.  I know this has to be a global setting in CUCM, as all phones do this, but I can't figure out where to go to change this.  We have recently extended the hours the office is open, so I need to change this.  Does anyone know where this setting is?  Thanks!
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Is there a way to use google voice with a VoIP phone that has only Cat 5/6 port ?
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I am brand new to VLAN's so please excuse my ignorance. We have an AllWorx VoIP phone system that has been on our regular network for a few years. Our employees plug their phones into the network port at their desk then plug their laptops into their phone's (yes we have WiFi but wired is much more reliable). So each phone gets an IP from the DHCP Server then acts as a switch and each laptop gets an IP from the same DHCP server.  We are growing and starting to run out of DHCP addresses on our LAN. If I set up a VLAN for the phone system would employees still be able to plug into their phones and get an IP Address on our network?
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Got a small business needing a new switch, dusty warehouse environment, 16-24 port 10/100/1000, with POE for VOIP phones, doesn't need routing so it can be unmanaged layer 2 ,  Last but not least small business budget.  I would like some opinions (including why) on fairly durable switches that won't break the bank.
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When you use Microsoft PSTN calling - can you use your old Cisco 7945 phones provided you load them with the SIP firmware instead of the SCCP?
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Hello Friends.

for a VoIP project I have to install and deploy Skype for business for 50 users. they will going to use most of the skype for business features like:
-IM
-voice call
-file sharing

but video calls and conferences are NOT important and required very often.

my question is what hardware configuration should I use for my server (CPU.RAM.HDD.NIC). as they are on a low budget they ask me to run it on a PC.

tnx in advance for your opinions
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Hey all, i have an issue that external calls has noise, delayed audio from the external side. Internal calls are fine.
We recently changed from the Telstra supplied modem to a Draytek Modem. All ports have been opened up the same as they were on the Telstra one, all lines and SIP registered straight away, however i have not been able to resolve the noise.
As Draytek have alot more advanced firewall settings, QOS, I'm not sure what feature / settings i need to change to test.
We are running freepbx 2.11.

Thanks
Matt
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I have multiple cisco switches, from 2960 to 3750, and we are using voip phones that use the same ports as the computers.
So I'm thinking to to leave the computers on the default vlan, which is vlan 1, and have the voip phones on vlan 200 or some other vlan.  As far as I know, to have each port in two separate vlans, I would have to make all ports trunk ports, is there a better or another way than doing that?
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I use Audacity for Mac,

Thanks
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I have an old CallManager (4.3). it works great and no one wants to upgrade it. I have several small offices and individuals working from home offices and in order to have working phones in their locations I have to do site-site VPN's to each location.
Is there way to create some port forwarding and avoid VPN? Which ports? Any downsides?
The firewall is Cisco ASA5510 and they have Cisco 7941 and 7970 phones if that matters.
Thanks!
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Hello, I'm curious, hence I ask you; Is there such a "recent college engineer", either software or hardware, or one that studied both, who would meet all of these requirements:
embedded Linux based data telecommunications system/ VoIP/SIP
satellite network application and
embedded software/ Firmware/embedded systems
automation scripting
Python
Java
Linux operating system/Linux Network expertise
Unix shell scripting
WEB GUI test automation
DO-178B experience
Satellite communication experience
Telecommunication experience/PBX switching systems

Any comments?
Do companies expect to find an "expert" engineer with all this knowledge?
thanks
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If you are using MS Lync to make a phone call to the PSTN and there are multiple SIP trunks to the PSTN (actually these are trunks to Cisco CUCM and then on to PSTN from there) - how does Lync decide which SIP Trunk to use?
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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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Hi,

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?

Thanks,
ELM-Server-Error.jpg
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Hi,

Running CUCM version 9.1.1 and I'm seeing a lot of reverse lookups, they are failing because my AD server is not setup to accept those but what I wonder is it normal to see so many? what causes the CUCM to execute these queries? I can see like 2 million request in the last 8 hours. You can see attached a few examples.

Thanks,
CUCM-queries.jpg
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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?
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Please provide me with a list of services similar to Magic Jack that allow calls to be forwarded to another phone number.
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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.
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I'm trying out a few different UK network sim cards.  For EE under the Enable 4G setting it offers, OFF, Voice & Data and Data only.

What has voice got to do with 4G because the other network sims dont mention Voice and data only data.
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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.
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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?

Thanks
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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 ...
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Is it possible to have a call attendant on Google voice , just even greeting or music?
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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).