Expiring Today—Celebrate National IT Professionals Day with 3 months of free Premium Membership. Use Code ITDAY17

x

Telecommunications

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.

Share tech news, updates, or what's on your mind.

Sign up to Post

specifically looking to send a message to
china

+86 186 1156 98xx
0
Meet the Family that is Made for Collaboration
LVL 2
Meet the Family that is Made for Collaboration

The TeamConnect Family product group as part of the Sennheiser for Business Portfolio comprising high-quality, technically well-conceived meeting solutions for business communication – designed for any meeting room and any meeting situation.

I am looking for a free cloud-based call center software such as Fenero. There is nothing wrong with Fenero except that they don't provide any support at all getting started unless you sign up to pay significantly and monthly.

In their comparison at the bottom of this page, http://www.fenero.com/, they mention Five9 and InContact as being more costly solutions. SalesForce is very costly and is out of the question.

We will be making outbound calls and doing emails.

Free or pay-per-call or pay-per-support incident would be a better model getting started than would a monthly option.

Assistance in finding a better solution for this would be greatly appreciated.
0
I don't much about the Avaya IP-Office 500.  I've attached some pics that may help.  The first pic is the blue cable that connects to the phone but is no longer providing service.  
During our rack equipment move, we may have re plugged in one cable on the IP Office in the wrong port.  I'm not sure if that matters as I'm not proficient in PBX.  I had one of the staff members mark cables based on what phone goes out if we unplug it.  I'm not sure if that's done yet.  I left for another meeting.  
Anyway, are the patch cables placement very fickle for the IP-Office?  We may have accidentally plugged in one cable to the wrong port when moving things and that caused the phone to no longer light up.   Sorry, I'm walking blind here.  I would greatly appreciate some feedback.
blue_cable_from_face_plate_to_phone.jpg
a_and_b_drops.jpg
IP-Office_Ports.jpg
0
For the Current PBX system, the Telco delivered an Adtran, off the Adtran there is an amphenol cable that is punched down to a 66 block.  There is also a CAT5 off the adtran that is also punched down to a 66 block.  

I am installing a Cisco VoIP solution,  but I am not sure how to connect my Cisco voice gateway router with a PRI/T1 card to the Adtran.  I'm not sure if the telco will have to hand the circuit off to me another way, or if I can just connect to the adtran using the ethernet port.  I'm having a lot of trouble getting info from telco.    Could anyone lend me some insight on this and their experiences?   I do have some VOIP experience, but no experience with traditional pbx, 66 blocks, etc.  Thanks.
0
The network provider in my area recently shut down their 2G service and now my GSM modem doesn't work. Is there another kind of modem that would be compatible with a 3G network? I've looked a bunch of other places and it doesn't really seem to be terribly clear about it.
0
Hello Everyone

I'm currently in the process of migrating our current PBX system away from asterisk to Freeswitch. I am using FusionPBX on Debian 8. I am using the freeswitch webapi to originate calls. I am at the stage where when I execute the command, it rings the call centre agents phone and the customer automatically without the agent manually dialling the number. I would like the ability to manually specify a caller id number for the outbound leg of the call. At the moment it is not sending any caller ID. I can manually specify a caller ID number in the extensions page, and it works statically, however we have a need for the caller ID to be dynamic.

http://X.X.X.X:8080/webapi/originate?{click_to_call=true,origination_caller_id_name='Click to Call',origination_caller_id_number=1000,instant_ringback=true,ringback=\'%(400,200,400,450);%(400,2200,400,450)\',presence_id=630@X.X.X.X,call_direction=outbound,sip_auto_answer=true,domain_uuid=52b92yy9-7fb7-52ae-9e9e0595058bcdaa,domain_name=X.X.X.X}user/630@X.X.X.X &transfer('SOME EXTERNAL NUMBER XML X.X.X.X')

What do i need to add to this web address to get it to send a custom caller ID number to the customer outbound?

Many thanks in advance.
0
Is there anyone out there IT Wise who has gone thru an NBN migration for multiple clients?

Our area is in the midst of deployment, and a number of my clients are reporting getting calls that sound suspect.  
  • One client, a law firm, was called by someone saying they were their PABX provider, and that the firm needs to replace their PABX because it isn't NBN compatible.  The trouble is its only a few months old and I would have thought any PABX sold would be NBN ready
  • Another was called by someone from their telco saying the same "you need a new unit" however the customer luckily has notes from the negotiation of contracts showing the PABX has NBN components already.
  • Another called by their telco, and has 6 month old system, was told by the telco they have no records of how the gear was configured or what's fitted in the PABX box, so the customer has to foot the bill for a new PABX

Questions
  • How are you helping clients assess what gear they already possess is NBN capable
  • Are there any resources you can recommend that can be used.
  • What has been your experience with the NBN rollout in your area
0
the system was setup with an auto attendant and 3 call groups for voice mail and after the auto attendant the caller would choose 1 , 2 or 3  and that would send you to the proper mail box and then you would hear that message. i can not figure out how to rerecord the auto attendant message or change the message for the mail boxes that auto attendant send the caller to. i can do intercom 500 from every extension and that takes me to voice mail management, but I can not figure out how to manage another mail box other than the extension I am on or how to change the recording on auto attendant. I can access both systems with my laptop.
0
I want to know about Best call center campaigns providing company of America with lots of free campaigns for off shore call centers.
0
Hello all,

we are currently using IVR designer with admin rights. Our corp is removing admin rights on the machines. Is there any way to user Avaya IVR designer without admin rights on machines?

Regards
0
New feature and membership benefit!
LVL 10
New feature and membership benefit!

New feature! Upgrade and increase expert visibility of your issues with Priority Questions.

This has just recently started happening.
On making or receiving a call my phone is switched to loudspeaker. The loudspeaker icon Is greyed out. Clicking the icon turns it green and the phone is still on loudspeaker. A second click greys the icon again and the phone reverts to normal operation. It's becoming annoying having to switch it every time.
I have moved a whole load of apps to the sd card, but I haven't loaded anything new recently.
Can anyone suggest how I can stop it being on loudspeaker for every call please?
0
How do you get a Cisco 3CX SPA514G phone out of Do Not Disturb mode?
0
When ever i use skype for basic chat the output on the screen shows first in non-italic characters and then repeats itself in italic characters.  Is there any way to shut this off and have it all appear as just one non-italic output?
0
Any idea how to fix this error?
0
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
!
boot-start-marker
boot-end-marker
!
vrf definition Mgmt-intf
 !
 address-family ipv4
 exit-address-family
 !
 address-family ipv6
 exit-address-family
!
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
 h323
  call start slow
 sip
  bind …
0
Outlook webmail account at Microsoft. Live  won't receive the two step verify codes in msm method ?
0
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
http://www.dbltek.com/8-Channels-GSM-Gateway-pd6563436.html
UCM6202.png
Goip1.png
0
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?
0
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

cheers
0
[Video] Oticon Case Study
LVL 2
[Video] Oticon Case Study

Open office environments can create the dynamics for innovation, but they also bring some challenges. With over 1,000 employees in an open office, Oticon needed a solution that would preserve the environment while mitigating disruptive background noises.

Watch how they did it.

I have twilio phone no. And just want to use at home using my home network is it possible?
0
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.
0
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.
0
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??

Thanks!!
0
Hello-

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
0
I currently use voip.ms, however their quality is lacking.
0

Telecommunications

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.