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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.

Hello I'm using CUCM 11.5 and using SIP phones. I have my DID's and looks like I have what I need to have setup so that I can connect to my ISTP. but when they dial in, they cannot establish a connection to download their config files.

I wanted to know what else do I need to setup in CUCM? For SIP phones?

We have CUCM V12.5 installed. I have users in London, Paris and Singapore and I am having issues with Call Forwarding.

I am based in London and if I put my 8851 on call forward to my mobile anyone in the London office can call my number and it will forward to my mobile fine.

If someone in my Paris or Singapore offices calls my number they get "Fast Busy". If I turn call forwarding off they can call me fine.

If I forward my number to another internal number it also works fine. The issue just seems to be forwarding to external numbers.

Each location uses its own Device Pools, Partitions and CSS's etc...

My guess is the call is coming to london and then trying to break out of the London Voice Gateway but the format is incorrect. The London Voice Gateway is a Cisco ISR4321 andis attached to a SIP line.

Thanks in advance
I'm setting up a 3CX PBX on Amazon Lightsail, and I'm having trouble with setting up conference calling that will allow external participants to dial in (like FreeConferenceCall.com, or other similar services).

I have my inbound and outbound calls working, so I assume my basic setup is okay. I have one number I purchased from Skyetel (the VOIP provider I'm using), and I have another number that's being ported in from Skype (not yet active on Skyetel).

However, I'm not sure how to properly setup conference calling on 3CX. I have a single extension (00), and I have all of my trunks (inbound and outbound) set with the "Trunk" value of *1949 (the last 4 digits of the number I purchased from Skyetel).

In Settings >> General >> Conference, I set my Conference Extension to 00 (the only one I have), and I've set my External Number to the number I purchased from Skyetel (the one ending in 1949).

Are there other settings I need to host conference calls?
Now that Skype no longer allows a customized voicemail greeting, I'm looking for an alternative.

I have a Skype Number that I'll port over to the new system, so it needs to accept incoming calls on that number. Other than that all I really care about is voicemail. Skype cost me about $100 a year for the number + my plan, and I'd sure like to stick around that price range.

Any ideas?
I have a Cisco Voice Gateway 4331 that handles all of our calls in conjunction with Cisco Call Manager.  The voice gateway has a PRI circuit connected to a port and three POTS lines using the remaining three ports.  In this example, I want to have the internal extension 3337 use the specific port of a POTS line on port 0/2/2.  This is for a fax machine (attached to CUCM via an ATA 190) that only sends and I am having trouble with it being reliable over the PRI.  I was hoping to tie it to a POTS line to avoid trouble.

I tried the following and it did not seem to work correctly.  I feel like I am missing an important component:

dial-peer voice 3199 pots
desc ***** Send faxes over pots lines for mailroom *****
preference 1
answer-address 3337
port 0/2/2
forward-digits all

In the above, I am attempting to identify the internal extension of the fax machine (3337) so that it can be directed to use the POTS line on port 0/2/2.  Is there another set of commands that I might be missing?
I've recorded a conversation using my cell phone and the built-in mic on my Acer Windows 10 laptop. My guest sounds, as you would expect, like he's on a telephone. But the "phone" quality is in many cases raspy. I can usually tweak and edit silence and such pretty good, but I need help here.

I've tried several different settings but just can't get my guest's audio to sound less sharp, less "telephone-y" and simply more clear.

Any audio production experts out there who can give me what I hope will be a little simple guidance to clean this audio up?
We just had an Aiphone IX system installed.   We assigned static IPs to all the devices.   We have a Samsung tablet and phone that we use to control the gates or speak to people when they push the intercom.

I have verified that the tablet and phone are set to Static.  Every day each of those devices changes the IP address.  We set it in the morning and later in the day they get a new IP.

Note:   Our DHCP has a scope of 50-200.   The IPs we have assigned are above 200.   One suggestion that we have not tried yet is to create a Reservation for those two devices.  Any thoughts on this?

I have discussed this with 4 technicians and all of them state that if an IP is assigned as static it can only be changed manually.  

We have been researching this with the company that installed it and with Aiphone.  No one can figure this out.

So my question is how is it possible for the IPs to be changing?   I know that it is not being done by anyone.   The devices are turned on in the morning and just sit on the desk until someone comes to the door or pushes the intercom.   Any ideas will be appreciated.
For years I have used Plantronics Supra binaural headsets along with the matching Plantronics headset amplifier/interface M10 or MX10. ( I am not at that location now.) I have the requirement that my phone audio be crystal clear at all times. I have never had any problem with clarity until yesterday. I never had considered VOIP because my internet speed was not ideal. Several months ago I got fiber and my up and down speed is 1 GIG with pings at 2ms. With this super speed I thought that VOIP would be an acceptable choice since it would save me more than 50% of my phone bill. Now that it is installed I am told that my transmitted voice is somewhat distorted and there is some sort of slight crackling in the background. I cannot live with this problem. I spoke to level one of tech support last night and he confirmed that I was indeed distorted. I have another line that utilizes the MagicJack. I phoned the tech on that line and the distortion was still present. I also switched from my headset and amp combo to a regular phone on the business phone system (Avaya Partner) and the distortion is still there. Level two is supposed to get back to me today and begin troubleshooting the problem. I was just wondering if any Expert has encountered this difficulty before? With such incredibly high isp speed the is the last thing that I had expected. If an Expert has any ideas please let me know.  

Configuration wise: On the isp's router there are two phone jacks. I go from jack One …
The phone is Polycom VVX 350, provisioned by RingCentral.

Is it possible to somehow program the Polycom phone to produce a distinctive ring if a specific number calls.  A mobile phone can do this.  I wondered if there is any way that I can get this feature? If not Polycom, is there another brand of VOIP desk phone that has this ability?

I am replacing a Cisco 2960 switch with a 9200. After copying & pasting the run config from the 2960 the interfaces are missing the following commands;

srr-queue bandwidth share 1 30 35 5
priority-queue out
mls qos trust device cisco-phone
mls qos trust cos

I also see that in our AUTOQOS policy-map for our Cisco phones that the 'police...….' configuration line detail has not pasted across

I am wondering if these commands are handled differently in the 9200 and have to be reconfigured accordingly.

Expert advice on this would be greatly appreciated.
I am using Freepbx 14 and working fine but I got thousands of attacks and in Intrusion Detection, my public ip  has been blocked sometimes and because of this calls are not working. I am using fortigate firewall and opened the 5060 to 20000 ports for the FreePBX so My question is 1. are ports forward mandatory for inbound route ( if I change the sip registration port from 5060 to other and do same with the trunk provider ) . Please let me know how I can make this FreePBX more secure so call disturbance would not occurred in future.
The user has moved into a new apartment.  They brought their Panasonic wireless home phone system, with a base station and two satellite phones with them.  

They would like to just give everybody they know only their cell phone number.  They need an adapter that would transfer all the cell phone calls to their Panasonic system.  This way they have the convenience of the Panasonic stations, and don't have to carry around a cell phone all day.

What is a good unit that will do this?

The user lives in an apartment complex that provides phone service as part of the rent. They were able to connect their Panasonic wireless home phone with 3 stations to this system.

However, even though the wireless phone receives calls, the caller ID isn't working.  Most inbound calls show the ID as DID/DOD.  Outbound calls show the town or just the phone number.  I called my iPhone, and my iPhone recognized the number as being on my contact list, and thus showed the name.

In addition, the answering machine built into the main station usually doesn't pick up.  Management said that an answering machine should work.  The Panasonic HAS an answering machine.

Is there anything we can do to get the caller ID and answering machine portion of the Panasonic phone working?

I am using PFsense2.4.4 with 3CX 16 and Everything (inbound and outbound calls) are  working fine but I am not able to register the phones over the VPN ( other end firewall is fortigate) I have done everything as https://www.3cx.com/docs/fortigate-firewall-configuration/  . The interesting part is I am able to work with softphone but not with IP phones( tested with yealink,polycom).
Someone spoofing my number and calling another person with mortgage offerings and other services.  
The area codes and prefixes of my number and other recepient's number are always the same.  Last four digits are always different.  Both numbers are with AT&T carrier in US.

Anyone had experience how to deal with it?  
Any solution besides keep blocking them?  Can AT&T do something about it?
We have a yealink IP SIP phone that needs to connect from the outside.  We have set up the phone and tested it internally and it's good so we moved it externally.  

In the phone, we edited the account and put in the public IP of our router (a Sonicwall NSA4600) and waited for the phone to register.  It fails to register.

I've checked the ports and I DO have the right ports open on the Sonicwall but I've also read a lot of posts about people having trouble with SW and SIP phones.  So, after reading, I have made the changes suggested on the posts I've read and still no joy.  

I'm using https://www.yougetsignal.com/ to test 5060 and it reports that it's closed.

Does anyone have any insight into how to make the SW work well with the SIP phone?

PS:  An alternate port scanning tool tells me the port is filtered.  So, I'm looking up how to turn filtering off in the sonicwall for this service.


I am a beginner with 3CX, and unfortunately I'm having issues configuring my new Grandstream GXW4104 gateway. I can get outbound calls to work, but for the life of me I can't figure out how to get the inbound calls to work properly. I've spent 2 days trying different things, tried several configuration guides, read the forums, been all over the internet, but nothing seems to work. I did have 1 successful inbound call routed to an extension (immediately after completing a re-configuration), but was unable to make a 2nd inbound call - it just rings out.

Strangely, after a standard configuration (as per the 3CX guide), with no inbound rules, inbound calls will consistently go to the operator, which is by default sent to voicemail, as I have no phone set up for that extension. If I then create an inbound rule or change any settings, no inbound calls come through. Changing the settings back to how they were also results in no inbound calls! I can only get it to work again by doing the gateway configuration from scratch!

Apart from the situation above, looking at the 3CX activity logs shows no activity when an inbound call is placed (log set to Verbose). Within the GXW4104 web interface, it shows that the line has a call coming through, yet 3CX reports nothing. I have set up the syslog server in the GXW4104, and from my untrained eye, when a call comes in, it is spitting out heaps of data, yet 3CX activity shows nothing. The gateway status always has a green light …
I'd like to know how to be able to subnet a PepLink Balance one router for less than 50 users. More concretely I'd like to have the IP phones on a  separate subnet. How can I go about that?
Please see the attached Fixed Line Operator (Telkom) Voice Mail answering Service. There is a distinctive tone indicated on the telephone handset when picking it up to indicate a message is waiting. Is there a device available that can indicate that a message is waiting and automatically retrieve the message and play it for the subscriber?
 This would save the trouble of each time dialling the prescribed phone number to retrieve and messages. It would be useful if the Message Waiting Tone could be detected and displayed fro the user. It would be wonderful if the device could play back the messages through a loudspeaker. I would imagine this would be a programmable dialler of sorts. The device should be senior Citizen friendly.
Hi Experts,

I am having an issue with my freePBX and the registration with my Net2Phone provider.
I have a SIP trunk setup for our phone lines

Apparently the connection keeps dropping because the registration refresh is not matching the provider's one. My provider ask me to change the refresh from 45 to 60 but whatever settings I put in the Registration Settings does not change the setting.
Can someone help me fixing the registration so are calls never drops?
Apparently I need to change the refresh from 45sec to 60 and the expiration to be 120sec.


This is what shows under the Chan_Sip Registry:
Host                                    dnsmgr Username       Refresh State                Reg.Time                
siptrunk.net2phone.com:5060             Y      8764774091          45 Registered           Fri, 14 Jun 2019 10:06:52
1 SIP registrations.

and the attached file shows the registration settings

Registration setting
Hi We used to use term Call manager. Now we always say CUCM, whose long term is Cisco Unified Communication Manager.  My question is why we say Unified?
What is the best Opensource chat or team servers that you can install on your premises ? Something like Matrix (Synapse) , RocketChat or Zulip?

Something that supports Chat and PBX integration.
I have a Cisco 2900 Series Router which has a EHWIC 3 card installed for my phone system.  I would like to replace this with a Layer 3 Switch or a firewall that can sit on the edge for "firewall" stuff (Intrusion protection, RTDPI, GeoIP, Anti-malware, etc.) which will allow me to connect my Cisco 7942 phones and allow remote connection to our network.  I will then connect to our switch stack on the inside.  Any Suggestions?  If you have other network suggestion on how to do this that would be great too.
Hi, we have just had Avaya IP Office and Voicemail Pro installed on a Hyper-V Virtual Server (A Dell Poweredge R7040 with 32GB RAM Intel Xeon Silver 4109T 2Ghz (2 Processors). The Avaya virtual server has been running fine since Christmas but over Easter it crashed. The telecoms guys that support it say that for 1-100 users it should be at least a 4 Ghz Processor (we have 75 users) and 2-3 cores.

Now, I can shut it down tonight and give it an extra processor, but would we really need to upgrade the processors to faster ones? It's a fairly new server and I thought that the Xeon Silver 4109T although low Ghz are pretty good? Thanks for the help.
This is (hopefully) a dirt-simple question for the Avaya IP Office gurus, but over 30 minutes of Google searching has yielded no results for me.

I have an existing Voicemail Pro module ("VM:Support") that works perfectly, and I know how to redirect external incoming calls to the module, but I need to create a single extension (x5555) that points directly to the module.  How is this done on an IP Office 500 / Voicemail Pro system?

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