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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Hi Experts,
I installed CME (ISR 4321 IOS version 15.5), with 40 Phones 7821, 1 IP Phone 8841, another 8851, and 2 IP Conference 8831 using SIP Protocol
all of these SIP Phones work fine, but the call transfer doesn't work, I don't know if some config missing, you find below the sh run,
thank you for your help.
CME-EHM#sh run
Building configuration...

Current configuration : 10874 bytes
! No configuration change since last restart
version 15.5
service timestamps debug datetime msec
service timestamps log datetime msec
no platform punt-keepalive disable-kernel-core
hostname CME-EHM
vrf definition Mgmt-intf
 address-family ipv4
 address-family ipv6
card type e1 0 1
no aaa new-model
subscriber templating
multilink bundle-name authenticated
isdn switch-type primary-net5
crypto pki trustpoint TP-self-signed-246793832
 enrollment selfsigned
 subject-name cn=IOS-Self-Signed-Certificate-246793832
 revocation-check none
 rsakeypair TP-self-signed-246793832
crypto pki certificate chain TP-self-signed-246793832
 certificate self-signed 01
voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
  call start slow
  bind …
Optimum High-Definition Video Viewing and Control
Optimum High-Definition Video Viewing and Control

The ATEN VM0404HA 4x4 4K HDMI Matrix Switch supports 4K resolutions of UHD (3840 x 2160) and DCI (4096 x 2160) with refresh rates of 30 Hz (4:4:4) and 60 Hz (4:2:0). It is ideal for applications where the routing of 4K digital signals is required.

I have a phone within my organization, while on DND, if called on the outside direct number, it rings while it's on DND.  Does anyone know how to program it to forward to voicemail and not ring through while the phone is on DND.

We have a MITEL System
- DBPrograming Version
- CS%000SessMangr.exe
- Call Processing Vers
- Voice Processor Vers 630.11.87
My recent headphone was broken and now I was thinking to buy a new one. I visited few websites but now puzzled that from which one i buy...?
Outlook webmail account at Microsoft. Live  won't receive the two step verify codes in msm method ?
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.
I am using a grandstream IP PBX UCM6202 and GoIP8 as a VOIP Gateway.
when i dial a number in this pattern +960 7788340 from PBX the call goes out to GoIP gsm gateway but from the GoIP outgoing call has a some unknown character due to space in front of digit 7. for this reason call is not dialed. if there is no space then there is no issue with the call going out. i am using an iphone numbers stored in contacts by defaults has some spaces. is there a way to remove the space when dialled out from GoIP gateway
below is a link to GoIP
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 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?
Portable, direct connect server access
Portable, direct connect server access

The ATEN CV211 connects a laptop directly to any server allowing you instant access to perform data maintenance and local operations, for quick troubleshooting, updating, service and repair.

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.
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.

Learn how to optimize MySQL for your business need
Learn how to optimize MySQL for your business need

With the increasing importance of apps & networks in both business & personal interconnections, perfor. has become one of the key metrics of successful communication. This ebook is a hands-on business-case-driven guide to understanding MySQL query parameter tuning & database perf

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?
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
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?


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.