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IP Telephony

IP telephony (Internet Protocol telephony) is a general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). Using the Internet, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN. The challenge in IP telephony is to deliver the voice, fax, or video packets in a dependable flow to the user.

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We are using FreePBX, and have Cisco 525G2 phones, and we added a SPA500s sidecar, how do i configure extensions on it?
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I have a question regarding ShoreTel Director setting.
We have about 100 DID numbers and we use "Hunt Groups", "Routes Points", and other settings.

Recently, we receive a number of fax calls on our company main phone number.
For now, we check the history of calls and check on the internet for the company's phone number.
Then, we call the company to inform that it is not the fax number, but it is our company main phone number.

Is there any settings on ShoreTel Director to transfer the fax call to the correct fax number?

I appreciate if you can tell me step by step process for this setting.

We have a new issue on our phone system.  If we receive a call and try to transfer it to another internal number, but that person cannot take the call...if we go back to the original caller and try to transfer the call a 2nd time, transfer is not available.  This all of a sudden started happening.  We are on CM version  Thank you.
What is the process for adding a ten digit (area code + seven digit phone number) phone number to an Office 365 Skype for Business (Lync) account?
We supply a software application that is run on Remote Desktop servers (each client has their own Virtual Server).  One of the areas I am currently researching is to provide the ability for users to click on a "Call" button in our application, and dial the number (whilst connected to the RDP).  I have found a couple of solutions that will allow me to press the "Call" button, which will then dial the recipient's number (along with the user making the call), which in theory, should be ok - but I am yet to test it and am interested in all options.

For me, the perfect solution would be for users to be able to plug in a headset on their local laptop, run the application on the Remote Desktop and call - without impacting server performance in any way.  Or, if they have a phone system already in place - somehow dial through that.

Any pointers, products, suggestions - would be massively appreciated.
Do the Comcast Panasonic KX-TPA65 VOIP phones also have a network jack for plugging in a PC (in addition to plugging the phone into the network)?

We have a Polycom SoundPoint IP 350 and a 450 at a remote office that connect to a FreePBX 2.11/Asterisk 11.7 box in our main office. The two sites talk using a site-to-site VPN via our FortiGate 30E firewalls. When the two remote phones register across the VPN, everything works great and there are no problems. However, if the internet goes out at the remote site, which it does often, the remote phones will never try re-registering again. Even restarting the phones by power cycling them does not let them re-register again. The two things I've found that works is to either upgrade OR downgrade the firmware by 1 version, or to restart the firewall at the remote office.

I've updated the firmware on both Polycom phones to the latest versions, applied the latest firmware to both firewalls (we have other remote sites that do not have this issue) and made sure that SIP ALG and VOIP application control are disabled on both firewalls. I can consistently reproduce the issue by unplugging the modem (not the firewall) at the remote site and then plugging it back in. When internet comes back up, the phones will not be registered and will never try registering again until the firewall is restarted or firmware version is changed. I've also played around with registration expiration and timeout settings on the phones, but this doesn't seem to work, either.

I'm thinking this may be a FortiGate firewall issue, but it's strange that my other 3 remote sites (using the same …
I'm running IP office manager v8.1
for some reason the call forwarding has stopped working.

I have the system to forward calls at night by manually hitting the "Night forward" button, which is ext 298 and it is forwarded to an outside number.

when i hit forward and hit that number, the calls don't get forwarded and they're not ringing at office either. help?
got a missed calls from last two months, This evening I got a ring from +381 628302791 at 4:08.Whoever calling rings 1-2 times and disconnects immediately.
searched on google Which country code +381, It's from Serbia I shocked why I got calls from this area.
I would like to know, is this going to be a problem? Like hacking etc??
I will have a chance of interview for the subject job position.

Can you share with me what I should look up and prepare before the interview?
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Just want to get folks opinions.  Anyone using them?  Any feedback?
Trying to implement sparkboards in every new office and eliminate things like conference phones, polycoms, and all that legacy stuff.

Question -
company's IP phones/voice managed by 3rd party.  No PBX on-site.  
Is SIP Trunking still required for the calls or its just calling over the Internet?

 I am considering "converting cable operator provided phone service to VoIP phone service".
 Can you recommend a vendor and explain why you like them?

 I am aware that there are multiple players - RingCentral, Vonage, 8x8 ... etc.

Thank you for your input in advance.
Need a question answered being asked of me and our Cisco CUCM 11.1 phone system. I need to know if Cisco CUCM 11.1 will integrate with NetSuite CRM via TAPI and/or CTI. If so, any third party apps needed or can it be done within CUCM?
Thanks in advance!
I'm running CUCM 9 and Unity connection 9.  All screen's on my Cisco IP phones go dim (black) at 5pm.  I know this has to be a global setting in CUCM, as all phones do this, but I can't figure out where to go to change this.  We have recently extended the hours the office is open, so I need to change this.  Does anyone know where this setting is?  Thanks!
SIPp is a free SIP traffic generating tool for Linux.

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)

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:

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.

$ autoreconf -ivf
-bash: autoreconf: command not found


Installing SIPp
•      On Linux, SIPp is provided in the form of source code. You will need to…
When you use Microsoft PSTN calling - can you use your old Cisco 7945 phones provided you load them with the SIP firmware instead of the SCCP?
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.

Hello Experts,

I am in need of options to connect an analog phone to an old, Nortel PBX that is in another building with no physical, wired connection between the two buildings.  There IS network connectivity between the two buildings, and I've been told by our phone support \ vendor that there are devices available that would connect to the analog port on the PBX on one side, and to the phone itself on the other side, and then both devices would connect to the LAN to provide connectivity between the PBX and the phone.  Our vendor, however, has very little info on these.  I am writing to see if anyone here has any familiarity with this concept, and if so, can provide any recommendations or guidance on makes \ models they have used successfully in the past?

Thanks in advance for any help that can be offered,

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We have ISDN PRI line 30 channel terminated into Digium TE133 card. Sometimes when we make outbound call on any mobile number we got error "All circuits are busy now". Once we redial the same number the call connects.

 How to capture  debugs  ?
For some project, we are exploring Twilio product "Programmable voice" . The pricing for per minute calling on normal voip is 1.5 cents whereas for SIP it is 0.4 Cents.

Thats makes me wonder how SIP differs from normal voip? Can I practically implement SIP in a small office of 4/5 people ? Can some expert exaplain me in simple English (Without using technical terms).

Twilio pricing I referred is here https://www.twilio.com/voice/pricing
I have an old CallManager (4.3). it works great and no one wants to upgrade it. I have several small offices and individuals working from home offices and in order to have working phones in their locations I have to do site-site VPN's to each location.
Is there way to create some port forwarding and avoid VPN? Which ports? Any downsides?
The firewall is Cisco ASA5510 and they have Cisco 7941 and 7970 phones if that matters.
i have Asterisk server running on two servers as Active/Active with 1000 Clients.Now we have configured the Lync Server i need step by step Guidelines for moving the clients from Asterisk to the Microsoft Lync with same extensions and dial plan.
I am in need to check the best solution for IP Telephony to work with Microsoft Lync Solution.any idea with logical comments will be really appreciated

already running ASterisk but want to replace it.
If you are using MS Lync to make a phone call to the PSTN and there are multiple SIP trunks to the PSTN (actually these are trunks to Cisco CUCM and then on to PSTN from there) - how does Lync decide which SIP Trunk to use?

IP Telephony

IP telephony (Internet Protocol telephony) is a general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). Using the Internet, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN. The challenge in IP telephony is to deliver the voice, fax, or video packets in a dependable flow to the user.