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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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.

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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!
Difference between VOIP and SIP.  Could someone give me a practical description of what the main differences between VOIP and SIP are?


Cisco 7961 IP Phone cannot hear people talking on either end. All looks good in the switch and in call manager.
Cisco 7942 phone will not register/show up in the MAC Address table. Call Manager is good. Wall jack/switch port checked out as good. Cabling checks good. No errors in the switch log.
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.
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.
What options exist for porting a Google Voice phone number over to other services?

A couple of users who I support are interested in doing this and want to find out what other services they can port (transfer) their existing Google voice phone numbers too.
Have a Cisco IP phone. Transfer call to another user, and the user is not able to pick up the transfer.
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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, 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@ for seqno 2 (Critical Response)
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
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?
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?
What is the difference between these user licences? I am running CUCM 10.5 with roughly
300 CIPC users and three SIP trunks for inbound calls from a Call Center.

LIC-UCM-10X-ENHP-A      UC Manager-10.x Enh Plus Single User License      CON-ECMU-LICMEHPA
LIC-CUCM-10X-ENH-A      UC Manager-10.x Enhanced Single User License      CON-ECMU-LIC0ENHA
LIC-CUCM-10X-ESS-A      UC Manager-10.x Essential User License User      CON-ECMU-LIC0ESSA
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.

We have our Shoretel HQ server in City A and there are two other sites that are connected to it and all of their associated phone devices.
We are decommissioning the data center in City A so we are trying to find out the best way to migrate the HQ server to City B.
The current IP subnet that it is on cannot be easily migrated due to multiple MPLS networks and dependencies on that subnet in City A.

Is it possible to create a new VM in City B and set up the Shoretel HQ server there with a new IP address? I assume we will need to update the sites and their associated devices with this new IP, but would it require a manual effort or is there a simple way to do this?

This way we can easily decommission the HQ server in City A once we have everything connected to City B.

Has anyone done this before?

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.
Hi, we have a data network here with Cisco switches that we manage. Now, there is a also a VOIP vendor who his own switches in our network. (He didnt want to use ours and VLAN everything).

Both the networks are on different subnets. Now, we noticed all of a sudden PC's started getting IP's from the phone subnet...and it's wreaking havoc internally. I tried to manually trace all 200+ cables in the office to see if someone plugged a phone device into the data network, but no luck..

How else can i troubleshoot this from say a switch level?
I have problem with Cisco ATA 190. Incoming fax is not working. There is no issues with the Outgoing.

Call flow: Vendor==>GATEWAY==>CUCM==>Cisco ATA 190

did a debug and proper dial-peer are matching?
When I am calling externally and internally Fax beep sound can hear.
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?

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.

Cisco 8851 phone not detected into the network. When I plug the phone directly into the switch in registers just fine in call network. Back at my work station. I plug the computer into the corresponding port and it works fine. Plug the phone in, and it get PoE just fine, but cann't see the network. I have cleared port-security on the switch. The phone is set to recieve DHCP just fine. The settings seem to be correct as I can plug the phone directly into the switch and the phone registers in Call Manager just fine. Plug it into the corresponding port back at the work station, and no go.

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.