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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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Is it possible to have a call attendant on Google voice , just even greeting or music?
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My company wants to compare pricing on hiring a full time switchboard operator versus outsourcing to an answering service.  We have one published number for the company even though all departments have a DID line.  The company is against automated attendant so the current choices are 24/7 answering service versus hiring an operator.  The idea is that the company main number would be forwarded to the answering service who would take the call and answer as "companyname."  They would then transfer the call to a DID number at our company to complete the call.  Obviously, the answering service would have a directory of DID numbers to forward to.

I am looking for some answering services that people have tried and have been happy with.  I just want to get an idea on pricing since we'd be comparing it to hiring a full time employee.
I am an office worker and need to buy a headset very urgent but confused that which one I buy form this series and from which trusted online retailer. If anyone knows a retailer that can be trusted to buy my product so, please reply me back.
Please provide me with a URL address of the Sprint PCS add a phone program.

I know several users who all currently have a single Sprint PCS cell phone and are interested in finding the details of how they can add a second phone to their Spring PCS account using the current Sprint PCS promotion that we have heard is going on until the end of June of 2017.
Does anyone know why the switch slows down so much that the VOIP starts to have Jitter and after a reboot it works fine for a week again. The switch runs the PC's through the Phones. There are no VLAN's programmed at the moment. The switch is stock standard. Are there any settings I can change?
Hi guys,

I am a network engineer and I have a requirement where I was asked to design IP PBX system with  Analog phones
can I use cat6A cable, CAT6A patch panels or I have to consider 110block?

Can anyone please guide here, it's very urgent...........
I have the above phone trying to VPN with a Dell SonicWall TZ400. When I put in the VPN information, listed below, the phone fails and gives me error codes that Phase 2 no response. I will list the three error codes I also see, if anyone can point me in the right direction.


SonicWall VPN Settings:

Policy Type: Tunnel Interface
Authentication Method: IKE using Preshared Secret

IPsec Primary Gateway Name or Address:

IKE Authentication:

Local IKE ID: Domain Name
Peer IKE ID: Domain Name

IKE (Phase 1) Proposal:

Exchange: Aggressive Mod
DH Group: 2
Encryption: 3DES
Authentication: SHA1
Life Time: 28800

IPsec (Phase 2) Proposal:

Protocol: ESp
Encryption: 3DES
Authentication: SHA1
Enable Perfect Forward Secrecy: Checked
DH Group: 2
Life time: 28800

In advanced tab, the only thing checked is Keep Alive.


Server: 50.XX.XX.209
PSK: *****
IKE Parameters: DH2-3DES-SHA1
IPSEC Parameters: DH2-3DES-SHA1
VPN Start Mode: Boot

Password Type: N/A
Encapsulation: RFC
IKE Parameters: DH2-3DES-SHA1
IPSEC Parameters: DH2-3DES-SHA1

Copy TOS: No
File Srvr: Blank
QTest: Disable
Connectivity Check: Never


IKE Phase1 received notify
Error Code: 3997698:18
Module: NOTIFY:305

IKE Phase2 no response
Error code: 397700:0
Module: IKMPD:353

IKE Phase2 no response
Error code: 3997700:0
Module: IKECFG:1184
I am hoping to set up call forwarding on our Samsung officeserv

I have looked at all the call forwarding options, 60? but they do not work.  

I note that this method will not override any pre-set forward settings nor will it work for groups.

Does anyone have any idea on how to make sure these things are not happening so call forwarding works?

officeserv device manager v4.93

I have twilio phone no. And just want to use at home using my home network is it possible?
We are looking into Skype for Business as our VOIP system. SB is seems working OK on voice side but we are struggling to find solution for paging system for classroom.  We currently suing voip phone system as paging system and will be out once we remove our pbx system out of building.  Any suggestion? Thanks for all of you in advance.
Manage your data center from practically anywhere
Manage your data center from practically anywhere

The KN8164V features HD resolution of 1920 x 1200, FIPS 140-2 with level 1 security standards and virtual media transmissions at twice the speed. Built for reliability, the KN series provides local console and remote over IP access, ensuring 24/7 availability to all servers.

We use twilio and sonic firewall in the company for our voip phone system.
And the issue is packet is drop more frequency and do not know what is the issue.

Most of the drops came from inbound calls.

Twilio has no issue.
I am a newly enshrined Telco Admin for Clark County Washington and need some help modifying
the dialing plan on 8 NEC 2000 phone systems.  They are finally instituting 10 digit dialing here
in a few months and need to make the appropriate changes to the 2000's to accommodate this.

Is there anyone who would like to make a few $$ consulting or assisting in the changes necessary.
Or can someone recommend a vendor who still supports these boat anchors??


I have an interesting setup. We switched to a hosted voip solution but for our emergency analogue lines we purchased two fxs vega 3050g gateways.

I have successfully configured the dialplans between the two units without the freepbx box as an in between. It is such a simple setup to just have it dial our security desk the pbx was not needed.

We are getting ready to deploy but have two lingering issues. To route calls between interfaces and the two gateway devices you have to initiate the sip handler to apply codecs and handle advanced features like call waiting. So dialing by IP between devices or loopback on between interfaces.

The one limitation I am running into is only one call at a time will take place between gateway... almost like the port is busy or it can only route one call at a time between gateways. Similar issue happens when trying to use call waiting... the first call connects but when I initiate the second call I get one indicator tone then the call fails with a cause code 41 (temporary network error)

Is this a matter of me needing to use more than the default udp port 5060 or am I just missing something here? I'm not a phone guy so I have a slight learning deficit.

I have combed through sangomas support site and manuals and found nothing like this... their support has yet to offer any solutions (on almost any of the setup.. which is troubling) so I'm turning to the community.

Thank you in advance
Thank you considering this question:

I would like to use a single Google Voice account with one phone number which is attached to it's own email account, and a second phone number which is attached to it's own email account.

I would like to be able forward SMS message from the first phone number to the first email address, and forward SMS messages from the second phone number to the second email address.  Is this even possible?

I am hoping to use the Google Voice Android app to be able to make calls from either number.  Is any of this possible, and if so, how could I set that up?
I currently use, however their quality is lacking.
My Headset is malfunctioning for quite sometime and now I have decided to buy a new one. I visited many websites but couldn't manage to make a choice. If anyone knows about some awesome and outstanding Headset model and brand, please suggest me. I will really appreciate if someone can also suggest me a credible website where I can buy your suggested headset. Price  is not a priority to me but quality.
hi all, my company is going to switch to voice over ip phone system within 1 week. before it is switched, do you know there is any stress testing that we can test our network ensure we can handle all incoming phone calls and etc.?

We have around 50 to 70 voice over ip phones will be used.

I have a 5 user license of UC Desktop suite. How do I release licenses that appear to be hung from previous reboots. I've tried rebooting the phone system and the UC Desktop PC that runs the shared services.

Currently I can only use three before I get the out of license error.

We are still using Avaya IP Office 403 & 406 control units. Wondering if there is a version of Phone Manager & Softconsole that will work on Windows 10 and with the 403 & 406 control units?
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I am running Freepbx and the international carrier is now putting restriction to dial out with access code. Is anyway I can bypass this adding a configuration file in FREEPBX.

Thank you
Hi Guys,

Can anyone help me find the setting in the console program find the part that lets you broadcast the telephone number. Currently our clients get a blocked caller as the number, consiquently quite a few of them ignore the call. I would like to choose the number being displayed from one of our numbers.

Thanks in advance.
outbound calls are working fine but incoming calls are not working on same port .

i run debug vpm signals and the output is below :

*Mar 6 08:01:42.526: htsp_process_event: [0/0/0, FXOLS_WAIT_RING_MIN, E_HTSP_EVENT_TIMER]fxols_wait_ring_min_timer
*Mar 6 08:01:42.526: htsp_timer - 10000 msec
*Mar 6 08:01:43.494: htsp_process_event: [0/0/0, FXOLS_RINGING, E_DSP_SIG_0100]
*Mar 6 08:01:43.494: fxols_ringing_not
*Mar 6 08:01:43.494: htsp_timer_stop
*Mar 6 08:01:43.494: htsp_timer_stop3
*Mar 6 08:01:43.494: [0/0/0] htsp_stop_caller_id_rx. message length 0htsp_setup_ind
*Mar 6 08:01:43.494: [0/0/0] get_fxo_caller_id:Caller ID receive failed. parseCallerIDString:no data.
*Mar 6 08:01:43.494: [0/0/0] get_local_station_id calling num= calling name= calling time=03/06 11:01 orig called=
*Mar 6 08:01:43.494: [0/0/0] htsp_dsm_close_done
*Mar 6 08:01:43.494: htsp_process_event: [0/0/0, FXOLS_WAIT_SETUP_ACK, E_HTSP_SETUP_ACK]
*Mar 6 08:01:43.494: fxols_wait_setup_ack:
*Mar 6 08:01:43.494: htsp_timer - 6000 msec
*Mar 6 08:01:43.498: htsp_process_event: [0/0/0, FXOLS_PROCEEDING, E_HTSP_PROCEEDING]fxols_offhook_proc
*Mar 6 08:01:43.502: htsp_pre_connect_disconnect, cdb = 3C645D8 cause = 31

*Mar 6 08:01:43.502: htsp_process_event: [0/0/0, FXOLS_PROCEEDING, E_HTSP_PRE_CONN_DISC]
*Mar 6 08:01:44.502: htsp_process_event: [0/0/0, FXOLS_OFFHOOK, E_HTSP_RELEASE_REQ]fxols_offhook_release
*Mar 6 08:01:44.502: htsp_timer_stop
*Mar 6 08:01:44.502: htsp_timer_stop2
*Mar 6 …

We came from an environment where we gave our users Blackberry devices since they needed to make international calls. We have a corporate deal with most carriers we were in.

We want to switch to a BYOD model, but this calling aspect is proving difficult to solve.

I had an idea though -

Let's say UserA is in UK and wants to call UserB in the USA. Standard phonecalls will connect UserA to UserB directly, which is expensive.

However - as an organisation, we have a corporate voice network that connects UK to US. Is it possible for a solution there UserA calls UserB, but the call actually routes

UserA > UK Corporate Phone system > US Corporate Phone system > UserB

That means we only need to pay the carriers local calls?

I believe this is called TEHO (Tail End Hop Off) :

Is this possible using a mobile app? I've seen some solutions like the below but interested in people's thoughts

I have TDA200  I have made no restriction for Landline & Mobile and restrictions for International calls

I want to know...for dialing an international call it should be in such a way that they should be able to dial it with Code only. I want

to know the steps how to do it
I have a client who uses cloud hosted VOIP service provided by a respected uk based voip provider.

My client is based in a managed office where the network is managed by a 3rd party company. My clients has been experienceing intermittant issues with the phones and they have asked the network people to check that the firewall\router\vlan configuration is suitable  for voip traffic.

The network people have come back saying :
I have reviewed the data captured yesterday, the problem IP address 109.*.*.70 is registered to THE VOIP PROVIDER. This is passing tiny data fragments through the firewall which is flagged as a security threat as this method is also used to hack routers and firewalls, this throws up an alarm and is blocked.

They have suggested we ask the VOIP provider why this is happeneing.

I am not a networking or VOIP expert. Can someone explain more about this and perhaps suggest why these "tiny data fragments" might be occuring?





Articles & Videos



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.