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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.

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?
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.
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)
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.

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

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.
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?
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.  

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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.
We have Cisco Callmanagers and Unity Connections for voicemail.  Our problem is this:
We have a number, say 1234.  It is set so that when a user dials it and it is busy or not answered, it rolls to 5678, which if not answered or busy, forwards to voicemail.  

The problem is, if you dial 5678 it works properly, eventually going to the mailbox.  However, if you dial 1234, it does eventually roll to 5678, but if not picked up, it then goes to the system general message instead of the vm for 5678.
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.

when I call Santander bank using skype
"welcome to skytel system"

but I can call using a cell phone

Please explain why you think I am blocked
I am using Freepbx 13 and want to block outbound calls on 911 no only.
Dear Experts,

I need recommendations of an Open Source telephone system Solution.

I need to let go of an old Cisco Unity Express. There is no budget for that office.

Is there a reliable product out there that I can set up in a Server machine?

Thank you
The current Network setup is customer site connected one SIP trunk each in US and Europe respectively
over MPLS Network. The customer is asking for cross region resiliency in SIP Trunks, is it possible? I'm not sure
if inter continent trunking will cause any issues? Please provide pros and cons.
Were experiencing issues with the Shoretel VoIP breaking up.
This is occurring predominately at one of our remote sites, however the main site is being affect, if slightly less.
Internal calls from the remote site to head office along with external calls are frequently causing problems.

Each site has 100/100mb link,
Shoretel switch at both sites.
Director and E1k and Ingate at main site,

Diffserve 467 enabled on HP POE switches
dedicated vlan for voice in place across the sites
sites connected by site to site VPN


Calling on all Cisco CUBE Experts;
CUBE setup for SIP trunking that that talks to the provider's SBC missing SIP port (5060) in the SIP URI, can anyone shine light on why it is happening? Is there a tweak or hack
someone can suggest ? The IP address is coming fine, BTW.

Need to create a simple IVR (Interactive Voice Response) process on Linux.
Any one has referred to any call center that can allow my way to make outbound calls including using my five 9 system? I need some one to use my auto dialer to contact my customers. Thanks
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?

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