Telecommunication occurs when the exchange of information between two or more entities (communication) includes the use of technology. Communication technology uses channels to transmit information (as electrical signals), either over a physical medium (such as signal cables), or in the form of electromagnetic waves. The word is often used in its plural form, telecommunications, because it involves many different technologies.

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The phone is Polycom VVX 350, provisioned by RingCentral.

Is it possible to somehow program the Polycom phone to produce a distinctive ring if a specific number calls.  A mobile phone can do this.  I wondered if there is any way that I can get this feature? If not Polycom, is there another brand of VOIP desk phone that has this ability?

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I have Asterisk version Sometimes my recordings have the audio of the agent and the audio of the client desyncronized. I want to know how could this be resolved.

I tried put jitterbuffer in sip.conf, and I changed to

Best Regards.
Recently I have migrated the 3cx on-premises to Cloud and all Ext. are connected through SBC. All inbound and outbound calls are working fine except the voicemail....I am not getting any voicemail after migration and in log I got Main line SBC:Unavailable  .......Internal voicemail are working fine.
Someone spoofing my number and calling another person with mortgage offerings and other services.  
The area codes and prefixes of my number and other recepient's number are always the same.  Last four digits are always different.  Both numbers are with AT&T carrier in US.

Anyone had experience how to deal with it?  
Any solution besides keep blocking them?  Can AT&T do something about it?
Getting "SMTP - unexpected data received. An unexpected end of message response was received.|" every couple of minutes, just started happening a couple of days ago. We haven't changed the internal SMTP server or upgraded our IP500v2's firmware.....stumped!
My current setup is this- I use a Watchguard firewall.
Interface 0 is external.
Interface 1 is trusted-
Interface 2 is trusted-
There is a VPN to another office that is

Our phone system is
If I plug a phone into the .2 network the phone will connect up without an issue.
If I plug a phone into the .3 network the phone will NOT connect up.

I assume there needs to be a policy in place to get the two to talk. I am unsure of what the policy needs to be.
hi - its a general question trying to understand how phones/phonesystems works .

got a client who uses nec phone systems SV9100 and nec phones.

connections are made like,:
phones are connected with telephone cables and going to patch panels
from patchpanels - using ethernet cable connecting to another patchpanels which has extension numbers.
extension numbers are connected back to phone systemsSV9100 in digital station interface and single line interface . (black cables in pic)
from phone system SV9100- a voip port is connected to lan switch (grey cable in pic)
and got another device "one access"  it has a ethernet cable connecting to lan switch.

i knew am confusing. but just need general idea- what is digital station interface and single line interface in sV9100 and what one device device does in the network ???
I can remote desktop using the Microsoft RDP app on android phone

I can also remote desk top on my laptop with WiFi

However when I try and tether my laptop to phone I cannot remote desktop to any server (10 different servers in different data centres)

I've tried turning off win 10 firewall, same.issue

Any ideas?
Hello, I've a Mitel PBX  for our Main office (200 users) and I would like to see if there is any low end PBX controller (15 users) that can be considered  for a branch office to handle internal call and route only the outgoing calls via Main PABX.
Hi I'm trying to setup a user on Avaya IP Office 6.2 to send voice mail to email.  The emails work for other users, but this is a user that is taking over an existing extension, so I need to modify (I think) the existing extensions voice mail to email.

Thanks all
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Skype for business 2015 Yealink T48S | Trusted Certificate
I need assistance with phones we recently purchased, all T48S handsets with Skype for business firmware.
I have tried all 3 latest available firmware and stuck with this version as it offered a simpler login screen for users.
Phones register correctly for as long as the the trusted certificate is not present.
Periodically the handsets will populate with a CA certificate on line 1 even though everything is set to disabled below and then the users are unable to sign into the phones.


i did some googling and found this command but its only relevant for SKype for Business online

What is causing the phones to download the internal root domain CA certificate?
We have Grandstream GXP2170s at our work and they're connected to a single 3CX PBX system.

Our company actually has 3 sub-companies, and we're planning on a multi tenant PBX setup in order to segment the various entities.

However, some of the executives have daily operations on all 3 of the companies, and need to connect to the PBXs and SIPs/Trunks of all 3 companies.

Will this be possible to do to have the Grandstream GXP2170s connect to 3 different PBXs and SIP accounts within the same headset?

I have a Samsung Galaxy S9 phone.

On this phone is an app that needs to be open in order to get the most out of the app.

Is there a way to have certain apps on this phone open once the phone is powered on and started up?

NEC sv8100 UC desktop Issue where I cannot untick Hide BLF button for some extensions, is there a setting on Webpro that I need to change for these extensions?

I am wondering if anyone knows what cordless units are compatible with the NEC SV8100 phone system.

I need to replace a couple of cordless handsets but cannot find any, even used. There are plenty of corded DTL-12D-1 BK TEL DT300 units available but I require cordless. Ebay has some base sets but no accompanying hand unit. I'd like to get a couple of more years out of the system which is still working fine.

Sometimes SIP channels get "stuck" in Asterisk and cannot be hung up without  restarting Asterisk. FreePBX shows the channels in the "Active Channels"  list under the Asterisk Info "Channels" menu but the "Active Channel" count at the bottom of the list is less than the number of channels in the list.  When trying to hangup one of the "stuck" channels using the channel request hangup <channel> command from FreePBX Asterisk CLI we get a 'channel not found' error. Asterisk version 13.9.1, FreePBX
I wish to setup On-Premise IP-PBX Business on my local server using proxmox vm. While being able to sync with an on premise Crm like bitrix24

I was thinking of getting switchvox. Kindly advise.

Have the following Scenario and need your advice for the best technical solution to be used.

- We have customer who is working with Oil Shale
- Mine Dimension : 3 KM x 1.8 KM x 80 Meter Depth  
- Around 15 machines working in the mine
- Machines need to be connected to the internet to send data either WIFI or GSM.
- The Customer Offices has Internet connection (Microwave)
- Distance Between Offices and Mine is around 3 - 4 KM
- One the Mine Surface - There is GSM Data Coverage (Still weak) but as the machines go down it looses connection

Our Issue we need a internet coverage for the machines, what do you recommend ?

Thank You
We recently started noticing that a certain floor of a building on campus can only call 1 direction.  When either side tries to call the one side in the bottom of our building cannot hear anything that the other side is saying. This only happens to the 3905's on the first floor of a specific building. If you use a higher model phone on that floor it doesnt happen and if you bring a 3905 up a floor or anywhere else and it functions perfectly.  Strangely, this isn't consistent as some 3905 phones on that floor can call some phones in other areas of the building and there are no problems at all.  I am stumped.  Any suggestions would be appreciated!

Phone system is Cisco
Switches are Cisco 2960S-48fps-L models with IOS 15
Call Manager and Unity versions are 11.5.1
Phone affected on the first floor of the one building are Cisco 3905's

If any log files are needed I can provide those.
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Hi All, I am attempting to configure an AAPT IP based trunk in 3CX via a dedicated TPG SIP service, and am struggling to get it working.

What I really am after is examples of working AAPT trunk configurations that I can compare my set up to.

If anyone out there could provide some examples of correct trunk configuration, I would be extremely grateful.

Wireshark shows OPTIONS messages successfully hitting the phone system from TPG SIP server, and 200 OKAY messages being sent to TPG SIP server.


 - 3CX on premise
 - Wireshark shows OPTIONS being received and 200 OKAY being sent to/from TPG SIP IP
 - Dual WAN connection with dedicated SIP service on WAN2
 - NAT on WAN2 to local IP of 3CX
 - Static routes are configured to route required SIP traffic in/out via either WAN1 or WAN2 depending on what port is required

Kind regards,

Hey guys,

I’ve recently started a new role and I’ve noticed that the guy before me has configured his guest WiFi to use the internal DNS for name resolution....traditionally, the approach I’ve always taken is to direct all guest access off the LAN and to use the DNS of our ISP.  Surely this is a security risk? The guys here say it was done to control/manipulate and secure DNS - but I’m not seeing their point.  Can anyone shed some light?

Im wanting to host simple PABX for multiple custmers and not sure which vendor to go for
we will host about 40 PABX's each with around 5 phones attached
support failover we will have instance in 2 separate DataCentres's for redundany/failover
needs to have single SIP trunk to host for all the calling voice channels
need to be low cost around 1$-2$ per extension and scalable as well
meed to be simple to provision and reliable
will be using yealink phones

any recommendation would be great
One of the Shoretel Server Services showing RED. ShorewareCDRMigration-UPG
Does anyone knows about this service? It seems that server working fine. Thank you!
Dear Experts, I have a question related to telephony service. We are using IP PBX Grandstream UCM6510 with SIP trunking from The Provider.

So as my understanding, for example if our number is +AA 710xxxxx; I create a conference room in UCM6510 at ext 8888; then when customers want to join a conference room with us, they will call to +AA 710xxxxx, press 8888. Am I right? (AA is my country code)

But the Boss now have some customers in USA, UK,... and he wants his customers will call to USA, UK numbers, (for example: +1xxxxxxxxx; +44yyyyyyyy) respectively instead of our number (+AA 710xxxxx)  to join our conference room.

Is this feasible? Can you please suggest the solution? Many thanks!
Dear Wizards, is there any tool for simulating Grandstream UCM6510?

just like in Network we have GNS3, EVE-NG, Packet tracer,,...Can you please suggest? Many thanks!


Telecommunication occurs when the exchange of information between two or more entities (communication) includes the use of technology. Communication technology uses channels to transmit information (as electrical signals), either over a physical medium (such as signal cables), or in the form of electromagnetic waves. The word is often used in its plural form, telecommunications, because it involves many different technologies.

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