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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.

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Hi,

I have been searching for a matrix listing the security advantages and disadvantages of wi-fi vs Bluetooth vs cellular vs ethernet.  I searched for a specific technology I mostly find publications talking about the general connectivity differences using the technologies but nothing side by side like a matrix that includes all three measured against the same baseline. I tried using two different articles and list the information in a spreadsheet but the categories and the baselines used were not the same - problem. Do you know of any resource that may have this information in a matrix view that I can review.  

Thanks
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Exploring SQL Server 2016: Fundamentals
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Exploring SQL Server 2016: Fundamentals

Learn the fundamentals of Microsoft SQL Server, a relational database management system that stores and retrieves data when requested by other software applications.

Looking for a VoIP phone system.
This client currently has a PBX black box supporting over 50+ phones. They are renting 5 lines from a local teleco.
They are building a very modern 100K square foot facility and will be moving to there in the near future. I started researching a new phone system to replace the current one.
I am not a telephone guy, so I would like to hear your opinions. First of all, I guess VoIP is the way to go, correct? I am kind of an old school and did not trust VoIP at the beginning. But after using Skype calling long distance for years, I find it has better voice quality than the landline most of the time.
I heard of 3CX, a cloud based phone system with no need to run an in-house PBX hardware or software. If the PBX is virtually on the cloud, how can we connect the 5 rented lines from the teleco to the virtual PBX?    
What solution would you recommend based on your real world experience? Reliability, voice quality, easy management.

Thanks!!
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Hi All,

We currently have an issue with a new build at a remote site.

The overall voice network is fully working at other locations, however the new site is having issues with inbound calls from the PSTN. The phones at both ends (internal and external) will ring, however no audio is passed. The call remains open, but silent.

Calls work outbound from the site successfully. The CUCM/Cube are on the main site, where calls work fine. The remote site is connected to the main network over a site to site VPN.

The only difference between this and other sites is the allocated IP range. The Cisco phones on the remote site are all using public IP addresses, where the main network and other remote sites are utilising private address space.

Any thoughts or suggestions would be greatly recieved.

Many Thanks,

John
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Hi

Does anyone know what the state is with the Shoretel (mitel) director licences?  Particularly with extensions?  We need to add more and cannot find out much info about them.
Awaiting comms co to update me.
Same question regarding the phones.  I believe 212k are EoL.  We need the line keys so what options do we have?

Thanks
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NEC SL2100 phone system started behaving poorly a week ago Sunday.  Rebooted all network equipment, and nothing changed.  Reboot the phone system and everything worked for about 20 minutes.  Same cycle of events, about same time frame.  Next checked all network equipment for firmware upgrades.  Wireless bridge Engenius EnStationAC went from 2.x to 3.1.2.11 on both sides successfully.  Problem returned yet again, but now a delay in getting dial tone of about 3-10 seconds.  If phone hangs up and connects immediately, no delay happens.  A monitoring computer is the only non-phone device on the entire voice network.  It shows random packet loss to phones.  The phone problem was originally they begain restarting at random.  They are not all happening at one time.  The delay never happens to VOIP phone on the system side of wireless bridge.  Engenius has yet to comment on the issue.  One phone (Only one phone) peforms IGMP actions accoring to Wire Shark.  Removing that specific phone solved the problem for almost 2 hours.  Yet it eventually returned.  All the time I have brought another laptop in for testing and moving it to both sides of the bridge.  If I am on the system side, I perform ping tests to the phones on the far side of bridge.  If I am testing the phone system resources, I am on the opposite side of bridge.  I perform around 1,500 packets to 3 devices simultaneously each time I test.  No latency or packet loss ever.  I have introduced a new PoE switch on the system …
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Dear Experts, is this diagram correct?

voip.PNG
We have 500 users, intend to use both VoIP service and Analog Tel service for backup. THere are 17 analog numbers to keep, so we choose Gateway GSM1024; for VoIP we use Grandstream UCM6510. We have several questions, can you please suggest?

- Can we use both UCM6510 and GSM1024 at the same time?
- Can we configure so that 1 department will use Analog line (GSM1024); other departments will use VoIP (UCM6510)?
- If one of these 2 fails , can the IP phone automatically change to the other, so that telephone service will NOT be interrupted?  
- Where can we find the reference link and installation manual for these devices?

Many thanks!
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I have a data jack that is terminated; but, not properly labelled.  I am trying to use a tone generator to find the correct patch panel connection that I need to use and cross connect the ethernet correctly to the end point.

https://www.youtube.com/watch?v=JHRIiw3OR6Y

The problem is that this specific data jack is in a machine shop floor and when I activate the toner every patch panel connection is triggered in the room.  If I turn off the toner generator every patch panel is still sounding off in that room.

If I try the toner generator in different locations(other rooms) there is no problem.  The toner generator works as I expect it to.

What could be causing the toner tester to sound off in that 1 specific patch panel?  Could it be that the ethernet cables were not grounded correctly?
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Hello everyone,
We have a ShoreTel VoIP phone system and we would like to use it to page different zones from the ShoreTel desk sets.
We have a 3 zone Valcom Page Control Unit.
Our paging goal is this:
  1. Page outside only
  2. Page inside only
  3. Page outside and inside at the same time
Is there a solution to accomplish these scenarios?
Do we replace our current Valcom 3 zone paging control unit with a new paging control unit capable of accomplishing the 3 scenarios?
Thanks
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I have installed Google voice on my Sprint PCS iPhone X.

Now how can I make my Google Voice number replace my Sprint PCS iPhone X caller ID number so that everytime that I call someone my Google Voice (and not my regular Sprint PCS phone number) will display on all outgoing calls?

Where do I go to enable this setting?

I received the email below on June 1, 2018 and I'm not sure how this affects my Sprint PCS Google voice service:

We are contacting you to let you know that per previous communication, due to upcoming upgrades to Sprint’s network, Sprint will no longer be supporting the Google Voice with Sprint integration. Effective today your Google Voice integration with your Sprint phone number (xxx) xxx-xxxx has been disabled by Sprint.

Effective today:
All outgoing calls (including international calls) and texts will be made through Sprint at Sprint’s calling and texting rates, if applicable.
All new messages, calls, and voicemails sent from your Sprint phone will not be stored in Google Voice. You will still be able to see your messages, voicemail, and call history from before June 1, 2018 in Google Voice on your Sprint device. You can also export this data from your Google Voice account at takeout.google.com.
You won’t be able to use Google Voice-enabled capabilities such as call forwarding, voicemail transcription, spam detection, and other Google Voice features. These capabilities can be enabled from your Sprint device. Click here for more information …
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android 7
automatic call recorder
free version
app version 5.42.1

recordings were good then I noticed last recording was at a low volume
Maybe in a version upgrade
volume is lower which is very difficult to hear
If I pay for paid version, do you think this will fix.
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Amazon Web Services
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Amazon Web Services

Are you thinking about creating an Amazon Web Services account for your business? Not sure where to start? In this course you’ll get an overview of the history of AWS and take a tour of their user interface.

I have a remote extension (yealink handset) that is dropping outbound calls after 35 seconds. I have diagnosed through logging that the asterisk server is not receiving the proper ACK, and the reason for that is that the asterisk server is looking for the reply from 192.168.1.4, which is the local subnet of the remote phone. Obviously the asterisk server will never receive a reply from that address. What changes do I need to make so that the asterisk server is looking for replies from the WAN IP of the remote network? NAT is setup properly on both ends. For reference, this is CompletePBX from Xorcom that we're dealing with and a copy of the asterisk log error is below.

[2018-08-16 09:27:08] WARNING[5720] chan_sip.c: Retransmission timeout reached on transmission 405286438@192.168.1.4 for seqno 2 (Critical Response)
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A customer of mine is moving their people from Office A to Office B, but is leaving the servers at Office A.

There are 5 networks in 3 different security zones that need to exist at Office B for the users, but the companies procurement department has vetoed using a Layer 2 type service because it's not in the current contract with their Telco provider.  :/

Net 1 - 100mb - Security Zone 1
Net 2 - 100mb - Security  Zone 1
Net 3 - 100mb -  Security Zone 2
Net 4 - 10mb -  Security Zone 3
Net 5 - 10mb - Security  Zone 3

The nets are all class C.  All the systems at Office B are new, and having different IP addresses is (amazingly / apparently) not a problem.

The Telco provider has sold them a pair of diverse 300 MB Ethernet MPLS circuits with three private VPNs which will be connected to two new Cisco 4431's at Office A and Office B that will both connect to all 5 network switches at Office A.  These routers will not directly route any packets between the 5 nets / 3 zones (there are other existing routers and firewalls at Office A that will do those tasks if needed), they are purely for connectivity between Office A and Office B.

Within the order notes for the circuits, I see the wording "Each MPLS CE router will utilize the Multi-VRF feature to segment traffic by application.  A total of 3 VPN's will be utilized."

I have no problems with routing protocol features (EIGRP or BGP) on high availability networks, and although I have never configured it …
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I am converting to SIP at a company site.  I set up the fortigate 60d identical to another one of our company sites where SIP is already implemented.

The SIP Cloud provider tells me that the firewall is not "passing confirmation packets  the phone system (PBX) sends out when it detects an incoming call",
and this is causing their system to transfer the call to our failover number.

I have opened up all requested ports (TCP/UDP etc), configurd policy,and QOS to high priority- everything they suggested, and as I mentioned earlier, it is configured exactly like our other site's fw.

I suspect it is a configuration with one of their servers.  In any case, Logging for all events is enabled in the firewall policy.  How can I tell if the firewall is blocking/not passing back the confirmation packets they SIp provider mentions?
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When I call my cell phone using skype on my windows 10 computer
I see the caller id that I selected on skype.com
I am spoofing my phone number to display another phone number that I use but am not currently using because I am calling from a computer.

But when I call a big business the number that shows up is not mine and can not be called again.

How do I purchase the caller id system that a big business has.
Caller ID that does not work with a spoofed phone number.
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good day,

I have two switch both configured for vlan 869 (Voice). The phones connected to SW01 can get IP address and phones connected to SW02 cannot get IP address. Can someone assist where I made configuration mistake. This is my first config as I am learning the Cisco commands. Attached are my configurations for both switch.
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For people that own / manage 800 numbers for businesses:

Do you get charged for / pay for calls from payphones?  Years ago there was a charge - 26c I think it was - that people that own toll free numbers were charged that went to the pay phone owner.  Wonder if that's still in effect. And for bigger businesses, do they (the 800 number supplier) not charge the 800 number owner?

And 800 numbers - does the owner typically pay a per minute cost? Or has it moved like outgoing phone lines to a flat fee / month?

thanks!
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Users are being forced to upgrade to latest skype client, currently v8.25, but from what i can see the user is not able to increase the chat text size.   This is troublesome and a backwards step for particularity elderly people using 1080+ screens.  

Skype (version 7) for Windows will stop working soon. Update to the new Skype (version 8) for Windows today! Don’t miss out on exceptional HD video calling, 300 MB file transfers, the ability to tag contacts with @‍mentions and more. Upgrade to the latest version of Skype to seamlessly transfer your sign in information, contacts and chat history. For more details and help, visit this page.

Does anyone have a hack/workaround for the text size?   If Microsoft have not built the setting into the initial design I can't imagine they will be adding it anytime soon.

Thanks,
--Paul
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Is there a converter or way to utilize a very nice slightly used Panasonic phone system that does not have VOIP technology built in to the phone system. It's perfect for a small business. Bought it a few years ago as the VOIPs were coming in. Wish I had not now. Unfortunately I have a cell phone but would like to utilize the home phone system NOT VOIP again because its conveniently setup to two different work stations. So I was wondering if the phone system I have can still be utilized with a phone line though some kind of a converter or am I forced to purchase all new phone system that works with VOIP? Because I am under contract with AT&T cellular. I unfortunately cannot switch and AT&T does not have landline services in my area so that means I have to get all new services with another provider? It just mean redundancy I just do not want to put everything through my cell phone. when I have a super nice phone system in house. Your thoughts appreciated.
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Cisco IP Phone 7941 still trying to upgrade.

Physically took phone to TFTP Server and uploaded current OS software to the phone.

Everthing in Call Manager looks good.

Cleared port security on the switch.

Phone daisy chained to PC.

PC has good Internet/Network connectivity.
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Exploring SharePoint 2016
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Exploring SharePoint 2016

Explore SharePoint 2016, the web-based, collaborative platform that integrates with Microsoft Office to provide intranets, secure document management, and collaboration so you can develop your online and offline capabilities.

I rewired my house with CAT6, and have a problem terminating the jacks at the end.

I bought IDEAL CAT6 jacks, which i've never really used before.  

I spent several hours on the CAT6 jacks, and I found it to be very uncomfortable.  the inner core is a nuisance -- i know it's for the better(less attenuation?)
the wires fit into the jack very loosely and i could barely guide the 8 wires in order into the terminating jack.  Once I crimped it, it was fairly loose as well -- i could pull and without exerting so much effort pull the wire back out.

i wired it once, tested it -- it failed -- no signal whatsoever.
i didn't know what was wrong and re-wired it a seocnd time, and still uncomfortable -- and failed again.

i searched for documentation and the internet is innuntdated with info on the old system using CAT5/CAT5e, and many videos even claim to show how to terminate CAT6, but don't get into the sled and liner.

i tried once more with the sled and liner using a poor IDEAL schematic, and then gave up.

after a while, of frustration i thought why not just try the CAT5e jacks -- and in 5 minutes I did what took me 4 hours previously.  felt comfortable --slid in snug, crimped strong. and tested perfectly.

my question:

can anyone provide a detailed CAT6 patch termination document including whether or not the sled and liner are necessary, and any other specifactions ( i think the wires didn't even fit in the liner)

my second question:

is performance going…
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Hello,
I have Avaya ip office in branch a and Nortel (upgraded to avaya) in branch b
I was connect branch b  to branch a with billion ssl
no its working with mikrotik sstp connection.
avaya can ping to Nortel and Nortel can ping to Avaya but the sip is not working between them
how I can resolve that ?
is there any documentation about that ?
thanks.
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I am looking for a Program or Script to dial remote Hayes AT Modems and log results. We have a large number of Hayes compatible modems connected to Verizon POTS lines. Many of these modems are in locations where there are frequent service interruptions. We would like to automate testing the availability and logging results. Is anyone aware of how we can do this?
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Hi,
I have a Panasonic TDE phone systeme and one of my phones, a KX-DT343, cannot do External Call. It says "Restreint".  
I am in the PBX software and cannot find where a restriction could have been implemented.
In a https://panasonic.ca/pcs/operatinginstructions/kxts620-oi-fr.pdf phone manual I found that the Restrict fonction is activated ON the phone itself but on my kx-dt343s there is no option visible for this.
It can be accessible from the phone or from the PBX  I guess but where?
Tx!
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Would it be a lot of data loss from SM to MM?
We have fiber from ISP [10Gb] is 9/125 SM going from SC to LC then into our LC 50/125 MM then into SM switch network module with SFP.

Would I have a lot of issues with that setup?

 I also read some people are using mode conditioning cables.  Will that help a lot?
https://community.fs.com/blog/mode-conditioning-patch-cord-utilized-in-gigabit-ethernet-applications.html
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SIP Provider Review, I have quote from Access Point Inc, I've never hear from the company before and bit skeptical where all their servers are. I don't to want SIP server siting on West end of country when our building is at East End.  Is anyone who experience with them? any feed back? or SIP provider who does good what they do?
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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.