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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Hi All, I am attempting to configure an AAPT IP based trunk in 3CX via a dedicated TPG SIP service, and am struggling to get it working.

What I really am after is examples of working AAPT trunk configurations that I can compare my set up to.

If anyone out there could provide some examples of correct trunk configuration, I would be extremely grateful.

Wireshark shows OPTIONS messages successfully hitting the phone system from TPG SIP server, and 200 OKAY messages being sent to TPG SIP server.


 - 3CX on premise
 - Wireshark shows OPTIONS being received and 200 OKAY being sent to/from TPG SIP IP
 - Dual WAN connection with dedicated SIP service on WAN2
 - NAT on WAN2 to local IP of 3CX
 - Static routes are configured to route required SIP traffic in/out via either WAN1 or WAN2 depending on what port is required

Kind regards,

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I have a problem with an asterisk server:

I have a SIP trunk from Vodafone. When I call from another provider ( lets say Orange ) redirecting from Softphone to another extension works. When I call from the same provider ( Vodafone - my sip trunks use vodafone ) and try to redirect from the softphone to another extension, the call is intrerupting.

This is extension.conf related to extension number used for redirecting: (I masked the real number)

exten => _yyy268,1,Set(CALLFILENAME=${CALLERID(num)}_${EXTEN}-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)})
exten => _yyy268,n,MixMonitor(/var/inregistrari/in/${CALLFILENAME}.wav,b)
;exten => _yyy268,n,Goto(ivr-liber,9,1)
exten => _yyy268,n,GotoIfTime(18:00-23:59|mon-fri|*|*?time,5692,1)
exten => _yyy268,n,GotoIfTime(00:00-08:00|mon-fri|*|*?time,5692,1)
exten => _yyy268,n,Dial(SIP/268,20,tk)
exten => _yyy268,n,Dial(SIP/241,60,tk)
exten => _yyy268,n,Congestion()
exten => _yyy268,n,Hangup()

THis is sip.conf related to extension used for redirecting:



THis is sip.cinf related to the trunk used:

I'm wanting to setup QoS for Skype for Business Online. When I do a packet capture I see that the real time
ports that come into play are the UDP 50,000 - 59,999. The article below calls the 50,000 - 59,999 as optional.
Is there any way through group policy to tell skype to use the UDP 3478, 3479, 3480, 3481 only or at least
to prefer it? Marking all TCP/UDP 50,000-59,999 for EF classification seems pretty broad.

Please tell me I'm wrong:  When using S4B to call a business that has a telephone auto-attendant, our S4B dialpad works just fine.  However, if I and an employee call a business together in a S4B call, the dialpad buttons do not work.  MS tends to suggest that this is a known bug.  We're about to agree and leave it at that... and leave S4B.

But really?  What an obvious thing to need to do.  We REALLY need to do this to train our employees on calling clients, etc.

MS seems to be moving from S4B to Teams.  Teams seems to be entirely geared toward pre-scheduled meetings where all attendees have agreed to join.  This is not our need AT ALL.  We need on-the-fly ability to add a voice call to an existing voice call AND be able to punch a dialpad for auto-attendants.

Therefore, we're looking for economical (5 users or less) solutions for our VOIP needs.  MS Office 365 E3 (which we'll keep) runs us $20/month, but in addition, for Skype PTSN dialing we also need $12/month/user for Domestic Calling Plan and $8/month/user for Phone System.  Therefore, our phone system needs are $20/month/user.

Can someone recommend some economical VOIP solutions? Thank you, yes we've looked - but that process is EXTREMELY unproductive (i.e. false/misleading claims on websites, feature listings are incomplete, etc., etc.)

Thank you!
Hi Experts,

I am able to access the call manager in our organization, I have a phone device and I can see it under Device --> Phone but I want to know how an anolog phone with DID phone number  will connect to call manager using internal extension usually using the last 4 digits as internal ext,

If the product Type Tye says : Analog Phone , does that mean it is a analog phone.
Im wanting to host simple PABX for multiple custmers and not sure which vendor to go for
we will host about 40 PABX's each with around 5 phones attached
support failover we will have instance in 2 separate DataCentres's for redundany/failover
needs to have single SIP trunk to host for all the calling voice channels
need to be low cost around 1$-2$ per extension and scalable as well
meed to be simple to provision and reliable
will be using yealink phones

any recommendation would be great
One of the Shoretel Server Services showing RED. ShorewareCDRMigration-UPG
Does anyone knows about this service? It seems that server working fine. Thank you!
I have some new VoIP phones and for some reason they will not configure on my clients network, when i took them home they work perfectly. I tried Wiresharking on a hub to capture the traffic, however i am at a loss as to what it means of what is causing the issue. The DNS is our Win2012R2 server and this then forwards on to the public Google servers.
We have a Mitel IP phone system (MiVoice Business release 8.0 SP3) and use 5320e phones.

Is there a way/process that users can have a call on hold, and then still use the phone to make a page?
We have a shoretel system that is not under warranty support anymore that we have been asked to do some hopefully minimal support on as part of our day-to-day IT support for a client.

the client has 4 different locations with shoretel IP420/IP480 phones.
I am trying to move a phone from one site to another.....and I can not get the phone to show up in the list of available phones at the new site.
the phone has a static IP on the proper LAN - no VLAN.....i went over the phone menu settings with a working phone to make sure all the settings are correct on the phone.
the phone tries to download a configuration from the server - but says it failed downloading

what am I missing?  maybe licensing?
are there log files on the server side I can look at to see what failed?

as always - any help appreciated!
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we are looking for 800 phone no with api. meaning when someone call 800 phone no. we want api to be called immedately to www.myweb.com/phone no
then redirect to 310.222.2222

do you know which provider can. do it? just like something out of the box without too much coding involved
In Skype for Business 2015, are there any APIs that I can use to query if a user has Enterprise Voice enabled?

I need to be able to differentiate users that have calling capabilities vs users that do not. Maybe being able to query if a user has a phone number associated with them would suffice too.
Dear Experts, I have a question related to telephony service. We are using IP PBX Grandstream UCM6510 with SIP trunking from The Provider.

So as my understanding, for example if our number is +AA 710xxxxx; I create a conference room in UCM6510 at ext 8888; then when customers want to join a conference room with us, they will call to +AA 710xxxxx, press 8888. Am I right? (AA is my country code)

But the Boss now have some customers in USA, UK,... and he wants his customers will call to USA, UK numbers, (for example: +1xxxxxxxxx; +44yyyyyyyy) respectively instead of our number (+AA 710xxxxx)  to join our conference room.

Is this feasible? Can you please suggest the solution? Many thanks!
Dear Wizards, is there any tool for simulating Grandstream UCM6510?

just like in Network we have GNS3, EVE-NG, Packet tracer,,...Can you please suggest? Many thanks!
I have one DC  with DHCP and DNS all in one. I am trying to connect a phone but it does not get an IP from the DHCP, Rebooted the server still getting (The DHCP service failed to see a directory server for authorization) error.

The phone (Cisco IP phone SPA 504G)  just sits on utilization network.
All other devices get IP and the lease time is set to 1 day.  It is when I try and add a new phone.
In Skype for Business Server 2015, Is there a way to disable clients from retrieving location information from the LIS/Secondary LIS every time the client registers with the server? I want to prevent the client from performing a HELD request to get location information when it first registers to the server. Is this possible with the on-prem server?
Is it possible to disable the voice feature of the exchange's automated unified communications operator, without removing the transfer configuration to marked extensions directly?

For example, if a customer dials the IVR, "Thank you for calling PCH, if you know the extension number, mark it now, otherwise the menu is the following, to call sales, dial 1, support 1, administration 3"

When I disable the aforementioned option, the part of "If you know the extension number, check it now" does someone know how to avoid this problem?

I would appreciate your support.

I'm trying to figure out if Gotomeeting can have better audio quality than SkypefBusiness. I'm talking about voice calls, where someone calls in for a meeting over the phone. Could there be a difference considering that they are both conference bridges? Can they offer better audio quality somehow?

Thank you!
How do you do!
My problem about algorithm, I don't have idea with resolving this situation.
I have two server:
1. First is ESXi on HP ProLiant G6 (rack based) - I'am creating on this server Virtual Machines for management office computers and have second VM for PBX (it's FreePBX with SCCP module for management and creating extensions for Cisco IP phone 7942G).
2. Second is simple PC keys - I use this computer for FXO PCI module with FreePBX server software. I connected city analog RJ11 lines to FXO PCI.
My problem is that - how do I receive calls from the second server (with FXO) on Cisco IP Phone (which is connected to the first server using SCCP)?
I can connecting two FreePBX between themselves with trunking. But it's working only with SIP protocol. Because, FXO lines come with SIP, I can receive calls with softphone.
Thanks to everyone for replying.
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Ok, this is the last part to getting our site paging worked out.
I need to know how I can get our ShoreTel phones speakerphones to work with Valcom speakers when a page is sent out.
We have a ShoreTel system 14.2 director and Valcom v-2003A paging controller.  We need our system to page all phones and all Valcom speakers at the same time.
Can this be accomplished?
If this scenario cannot be accomplished with our current equipment then what would I need to replace/upgrade?
Hi all, we need to have call recording for our VOIP system, does any know of a 3rd party vendor who offers this? Our current vendor and we are not switching anytime soon.
We are transitioning to Skype for Business from an older product called Spark.  With Spark it is fairly easy for each user to populate their own
groups with contacts that are saved and used to IM thru skype.  How can we populate groups without adding them one user at a time. We have some fairly large
We have been using Skype for Business (formerly Lync) for IM and video conferencing but are thinking about using Skype PSTN as a replacement for our current PBX.

We are in the process of organising a trial for some users to pilot Skype for Business as their only phone option.

One of the things we need to do is gather some data once their trial has finished and I am looking for help identifying test criteria for the new system.

If you have any thoughts about the sort of questions we should be asking users to assess the suitability of Skype for Business, then please let me know.

Thoughts so far:-

Which microphone device do you normally use for S4B phone calls?
Which amplification device do you normally use for S4B phone calls?
Frequency of use
In a typical day, how likely are you to make a phonecall using S4B
In a typical day, how likely are you to receive a phonecall using S4B
Sound Quality
How does the sound quality compare to landline
How does it compare to mobile

Any questions / comments appreciated.

How can I troubleshoot a Cisco IP Communicator phone stuck at "Configuring IP" during boot up? I've configured the device and added an extension and restarted the tftp server on the subscriber. But registration appears stuck. Where can I go to see logging as to what's failing?
Ok, so Windows 10 made some changes and i can no longer use the installed TAPI Driver to initiate phone calls from my access database. After some research, i am able to perform the following.

By Clicking on the start button and typing run then pressing enter the run windows opens.  I then can type this command tel:97025551212 and the tel protocol will dial the number using our Voip phones.

I also can get the computer to dial our phones by going to a command prompt and entering Start tel:97025551212

I need to be able to do this function within Microsoft Access VBA.

Any ideas anyuone.

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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IP Telephony