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

SIPp is a free SIP traffic generating tool for Linux.
http://sipp.sourceforge.net/

SIPp user manual says you can install SIPp under CYGWIN on windows. However I am not experienced  with compiling applications to run under Linux and need help getting SIPp up and running under CYGWIN on a windows10 machine.

I have successfully installed CGYWIN and included the following packages (all successfully)
gcc-core
gcc-g++
gcc
libncurses
make

After the CGYWIN install, I put C:/cygwin64/bin in the win10 systems’ environment variable PATH – so far all ok and CGYWIN seems to be working fine.

In addition, the SIPp install instructions state:

Warning
SIPp compiles under CYGWIN on Windows, provided that you installed IPv6 extension for CYGWIN (http://win6.jp/Cygwin/), as well as libncurses and (optionally OpenSSL and WinPcap). SCTP is not currently supported.


QUESTION 1 -  Do you know what this is???    IPv6 extension for CYGWIN http://win6.jp/Cygwin/ 
is it a CYGWIN package, and entire install version??
What/how do I need to do to check/install?

QUESTION 2 – Nothing happens when I try to run “autoreconf -ivf” ...but this might have to do with Question 1 not being addressed yet.

 /cygdrive/c/Backup/tools/SIPp/3.3
$ autoreconf -ivf
-bash: autoreconf: command not found


+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++
FROM DOC


Installing SIPp
•      On Linux, SIPp is provided in the form of source code. You will need to…
0
VIDEO: THE CONCERTO CLOUD FOR HEALTHCARE
LVL 4
VIDEO: THE CONCERTO CLOUD FOR HEALTHCARE

Modern healthcare requires a modern cloud. View this brief video to understand how the Concerto Cloud for Healthcare can help your organization.

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
Hey all, i have an issue that external calls has noise, delayed audio from the external side. Internal calls are fine.
We recently changed from the Telstra supplied modem to a Draytek Modem. All ports have been opened up the same as they were on the Telstra one, all lines and SIP registered straight away, however i have not been able to resolve the noise.
As Draytek have alot more advanced firewall settings, QOS, I'm not sure what feature / settings i need to change to test.
We are running freepbx 2.11.

Thanks
Matt
0
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
On Demand Webinar: Networking for the Cloud Era
LVL 9
On Demand Webinar: Networking for the Cloud Era

Did you know SD-WANs can improve network connectivity? Check out this webinar to learn how an SD-WAN simplified, one-click tool can help you migrate and manage data in the cloud.

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
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
Technology Partners: We Want Your Opinion!
Technology Partners: We Want Your Opinion!

We value your feedback.

Take our survey and automatically be enter to win anyone of the following:
Yeti Cooler, Amazon eGift Card, and Movie eGift Card!

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

Thanks
0
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.

SV8100
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.