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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I have on VLAN 3 (10.10.110/24) a PBX and VLAN 4 ( IP phones.  routing between VLAN 1 and VLAN 2 is working.
On VLAN 1 ( I have computers, W2008 DC, W2008 database.
I can ping from VLAN 1 the PBX on VLAN 3
I have setup the DC IP address as NTP server on the PBX. but I m seeing the status disconnected. That's probably mean from VLAN 3 I can"t reach the DC located on the VLAN 1

I am trying to find out how I can fix this to be able to synchronize the PBX and IP phones clock with the DC.

Any idea?
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Our chairman calls his assistant and dictates emails over the phone for his newly hired assistant to type and send out. Sometimes she has difficulties to keep up with him and we really need a device, program or something for her to record his conversation and listen to again ?
Here's an easy one for ya...  I've done this before in Google Voice but I can't seem to find the feature to enable forwarding to our VoIP system for whatever reason
call 4 phone numbers at the same time.
Only 4 not 400. Not looking for an enterprise solution.

Maybe something that combines with google voice or skype.
Want to use windows 10 computer and regular headset.
Ideally no additional hardware but if hardware is needed for 4 phone numbers then
Skype sound not working.  The sound works on other applications no problem. I've checked the device settings - done the audio test - has worked fine. Am running both Skype for business and "ordinary" Skype but they've lived happily together until now.  Try the Sound Echo test - not a peep out of it...any suggestions. Thanks.  Skype version 12.1803.279.0
We're being told that shoretel 14.2 will be EoL in the near future.  We're also being told that we will have to pay for the upgrade to connect.  Is this correct?  We have maintenance with shoretel and the supplier...
Also what issues are people experiencing with it?  I read that new communicator is pants..?
I am having some issues with some phones and was hoping someone could hopefully point me in the right direction. I am not a phone guy by any means, so excuse any mistakes or anything that is unclear. Our past set up was as follows

Site A - Sonicwall NSA 250 M with Avaya IP Office 8.1
Site B - Sonicwall TZ 205 with 20x Avaya 9608 phones

The sites are connected via a Site to Site VPN.

A week or so ago, we swapped out Firewalls. We moved Site A's to Site B, and put a Sonicwall NSA 2600 at Site B. We did a simple export/import of configs. Even though they were different Firewall models, Sonicwall documentation said it was supported, and we haven't had any issues. Except one.

Our phones seem to experience call dropping and quality issues. We get 10x dropped calls a day, and inside IP Office I can see Quality of Service Alarms going off like crazy.

I have set up QoS and BWM on both sides of the Firewalls, I don't believe bandwidth is the issue.  It's ONLY my remote phones at Site B, which are all H.323 phones. But if someone from Site A calls Site B, there is a chance it will drop as well. Site A can call Site A all day, or externally, no issues. I played around with H323 transformations on the Sonicwall, and that actually seemed to fix the issue, but after enabling it my phones would deregister themselves after a few hours, and would not re-register.

I have set up wireshark on both ends, nothing out of the ordinary, no increase of traffic when issues comes up. …
We have 4 remote sites connected via MPLS T1, with Shoretel SG30 switches at each site.
One site in particular has had network connectivity issues for a long time, and so I thought I'd look at the state of the switch
since they lose Shoretel Communicator connectivity several times/day, on their PC's.

In Shoretel Director,  in Alerts section, out of 100 SG30 "Switch has lost connection to the network" messages,  20 of them are at other sites(not experiencing issues)

Is this common for a Shoretel SG30, to display so many alerts for a switch?

e.g. here are the entries for yesterday: (displaying "Switch has lost connection to the network" message)

2/5/2018 23:19
2/5/2018 23:19
2/5/2018 23:19
2/5/2018 23:19
2/5/2018 23:19
2/5/2018 23:19
2/5/2018 22:58
2/5/2018 21:50
2/5/2018 20:40
2/5/2018 19:30
2/5/2018 18:26
2/5/2018 17:15
2/5/2018 17:13
2/5/2018 17:12
2/5/2018 16:07
2/5/2018 14:57
2/5/2018 13:50
2/5/2018 13:48
2/5/2018 12:42
2/5/2018 12:39
2/5/2018 11:33
2/5/2018 10:26
2/5/2018 10:23
2/5/2018 9:17
2/5/2018 9:16
2/5/2018 8:08
2/5/2018 6:59
2/5/2018 5:50
2/5/2018 3:35
2/5/2018 3:32
2/5/2018 2:27
2/5/2018 1:17

tty tdd accessibility for hearing impaired.

online how to set up phone calls voip that allow for tty tdd as a client

What hardware should be purchased.
I was working on upgrading some wireless Access Points and for some reason, two Polycomm Sound Track 7000 woudn't boot up normally and would show red LED status after it shows the Polycomm logo.  I googled and saw that one suggestion was hardset it by pressing 1,3,5,7, which I did but still the same issue.  We're trying to connect those phones to the original power adapter (if we can find it).  They are using PoE right now.

I connected a different Polycomm desktop phone to that jack and confirmed the port and cable are fine.  I even tried connect the IP Soundtrack 7000 directly to the switch port and has the same issue.  

It is weird that both IP Soundtrack 7000 would have the same issue.  Does anyone know what might be the issue other than to RMA it?

Also, what we noticed is that after the upgrade of our Wireless Access Points, the new Access Points wouldn't lid up until we changed the ethernet cable to a different switch ports.  This happened to about 4-5 Access Points and they are all PoE.

What would cause the original ports not to work and not sure if the Polycomm is related.
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Dear Experts, we are setting up an Asterisk IP PBX in an office environment in Ubuntu Server. What do we need to purchase so that internal user can make phone calls to outside and vice-versed?

Many thanks in advance!
Hello Team,

I want to track the missed call notification on Cslogger skype for business 2015 tool.

Please suggest how to trace it

we have skype onpremises 2015

We have a Switchvox PBX server and Digium Phones on the desktop.  I am interested to learn what other people are using when employees are out on the plant floor and internal communication needs to happen.  Currently, employee has to be paged and then they have to walk to a phone and dial the extension.  It would be nice if we could use a different solution to limit the amount of time it takes to answer a call or communicate with fellow employees on the plant floor.  Thoughts?
I need to add dial in numbers for conference calls for my organization within Office 365.

We also need to make sure that whenever a user schedules a Skype meeting within Outlook 2016 (through Office 365) that the dial in numbers will appear below the "Join Skype Meeting" section of the calendar invitation (see the screenshot).

How can this be done?

We have an Avaya IPOffice R9 System that is connected to a SIP Provider for outbound calls.  When some users call come customers the number is Identified at JPMorgan and not the Username that we have setup in the Avaya system.

If that same user calls my Home Phone I see the name that we have assigned the user not he Avaya SIP tab.  So I know its going though our SIP Provider just fine.

Is there some Caller ID Database that could be used for number lookup that would override the Caller ID Name that is sent along with the call?
I had this question after viewing phone calls and google voice.

on windows 10 desktop
I am looking for a browser addon that can record google voice phone calls

Related question is how to do on google server by pressing "4".
I am looking to install a small low cost WIRELESS VOIP phone system for my church.  Requirements are simple - VOIP phones that will operate via the Church's 802.11 wireless network.  We have 2 incoming analog telephone lines.  Will need 5-10 telephones with voicemail on each.  Would like a desktop style phone in most locations (vs. a small handheld).  Recommendations please.
I know a little about VOIP so please bear with me on this <g>

I know of services like Vonage that use an adapter - plug the ethernet into the box and it has an rj11 for a regular phone.

Then there are services that use a 'VOIP phone' like the grandstream GXP2130.  Is there a word or phrase that distinguishes these 2 type of services?

or most any service can use both - the adapter or voip phone?

And specifically, i have a spare GXP2130 phone. I'd like to know if I can set that up to make & receive calls using google voice?

or are there other free / very cheap services? I don't care about reliability really.  if its down for hours at a time not a big deal. not looking to port a number to the service. Just have another line / experiment with this gxp2130
for a small business which conference system will you recommend?

what are the leading brands out there?
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windows 10 chrome browser

Using google voice, I want to call a government worker.
Phone rings to voicemail.
This government worker does not call back if I leave voicemail

So I want to set up a way to call numerous times thoughout the day but disconnect if phone reaches voicemail
windows 10 chrome browser

google hangouts
"your call is free" dialog box
just an extra click if I make many phone calls

is there a way to make a phone call with less google voice clicks
windows 10 chrome browser

This extension does not have click to call

many pop ups but I could not figure how to make it work with google voice
many choices of integration but google voice not included

Is there a google voice click to call chrome extension so I dont need to copy and paste phone numbers to make phone call
I call an insurance company and there is no "press 1 for english" today but there was yesterday.
And there is a long hold which wasnt there yesterday.

Todays phone call (I tried on 2 phones) goes straight to hold music.

If I dont press 1 for english, how does company know I want to speak in english?
Are buttons set up by company just for "future reasons" and the same operator always picks up?
Hello Experts - Our company switched to the iPhone 6s and we've run into a problem.  The phone does not appear to be correctly passing number presses when dialing in to our phone system and attempting to use the auto attendant directory.  Instead, it will constantly say the name was not located.  It works fine when not using Apple Carplay from the same location.  Any ideas on what I can do to fix this?
Hello Experts,
We are using Vicidial version 2.6-381a with Asterisk 1.4.44-VICI, I have strange issue one of my user extension 508 voice mail going into ext 801 voicemail box when using DID but when dial extension 508 it goes into right mailbox of 508, i have checked the user phone settings, DID settings to make sure DID rings and send voicemail to right extension from vicidial gui, here is Dial plan for ext 508 which is setup properly can someone please help me figure this out. Thanks

exten => 3508,1,AGI(agi://
exten => 3508,n,AGI(agi-NVA_recording.agi,BOTH------Y---Y---Y)
exten => 3508,n,Wait,2
exten => 3508,n,Playback(/var/lib/asterisk/sounds/44048020)
exten => 3508,n,Wait,2
exten => 3508,n,Playback(/var/lib/asterisk/sounds/85100075)
exten => 3508,n,Wait,2
exten => 3508,n,Dial(SIP/508,15,Ttr)
exten => 3508,n,Wait,2
exten => 3508,n,Playback(/var/lib/asterisk/sounds/85100031)
exten => 3508,n,Dial(SIP/508&SIP/801&SIP/505&SIP/507&SIP/509,15,Ttr)
exten => 3508,n,Wait,2
exten => 3508,n,Playback(/var/spool/asterisk/voicemail/default/508/unavail)
exten => 3508,n,VoiceMail(508@default)

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.