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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VoIP - My customer reports that their phone lose connection to the hosted service every day.  They have to reset multiple times per day.

I will be onsite sometime tomorrow (April 11) - hoping to be able to access some expert assistance.  Meanwhile, if somebody could point me to a link that I'm sure exists, to help me in troubleshooting VoIP.  I'm good at networking, but have minimal experience in troubleshooting VoIP.  Thanks in advance.
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Dear experts,

I have Cisco voice gateway routers and I need to know if there are any fax lines active in this organization. Is there a way to figure it out and what is the process?
unfortunately, the company does not have enough info but they have many MFP printers with fax lines and are working.

Thank you
Hey all, one of our users (at least we think it is only one) is having issues with caller ID, note that we use skype for business/O365 online.

For example: When he calls me I see his proper skype/office # onĀ  the caller ID because he is calling me from his skype for business account....but if I miss the call, I do get an email notification....but in the details it shows as if the call came from his mobile #.

Any idea where the issue could be?

I did notice something in the desktop client settings...under Options - Phones, I do see his office and mobile #s listed, but am unable to change the mobile# because the "Mobile Phone" button is greyed out (see attached image)

I found this article that said its an admin setting, but I can't find it anywhere...again, not sure if this is the cause but figured I would mention it.


Hello Experts,

I at a customer site and they a VALCOM V-2006A Amplifier , I configured the SIP paging adapter and connected it to the Valcom V-2006 Amplifier, the SIP adapter has paging extension. Now the question is how the speaker should connect to this amplifier? What do I do to make this work?

I don't know much about the VALCOM V-2006A amplifier and I need this to work.

Thank you,
We are having a strange issue with our Skype for Business.
Its been running just fine for sometime now but we have some external clients that are having an issue with our Skype ids.
What's happening is that we can send them a skype message and they receive it, but when the try to reply back to us, it eventually times out for them, and we don't get their reply.
Also when they bring up their list of other clients including ours, ours show up as status, Presence Unknown
Does anyone know what I can check to see why our IDs would be appearing to them as Presence Unknown?
Our company planing to buy new telephone system. I was looking to buy Avaya cs1000. However, it become at end of life. We are looking to have PBX that can host less than 500 IP, land line, and Mobile phone with voice mail feature. We have old system Avaya BCM 450 and Nortel IP phone 1120 and 1140. Which system can be better chose to us ? and can connect to these phones? I want to use them beside buying some new IP phones.  

If its not possible to have good system to connect to these phones which PBX can be good and with reliable prices?

Cisco Unity voicemail to email logs

We have a Cisco phone system and it is set up to send the user an email each time they get a voicemail.
Some users reports they don't get the voicemail emails and I want to start troubleshooting it by looking at the email logs.
If I look at our O365 email logs, Message Trace, I don't see anything from the phone system going to any user. I know it works for me so I looked at the O365 email logs for me but there's nothing there from the phone system.

Where would I find the log that shows me the email with the voicemail attachment?

SMTP settings in Unity are set to Port 25 and the SMTP Domain is "domain"-cuc1."domain".us
The SmartHost is set to our internal SMTP relay (nothing in those logs either)

Grateful for any help!


We are using voip phone and provider is 3CX . Currently we are receiving a lot of phones call with unavailable number. 3rd party support is saying that everything is ok from their side. When we ask customer if they are ringing from withheld number , most of them say that they are not ringing from withheld number. Please advice what could be wrong
Using Microsoft Call Quality Dashboard I stumbled on the fact that call quality for people using 802.11ac wifi is 4x better than for someone on 802.11n radio.
What could account for such a dramatic difference?

For one month 802.11ac: 978 good calls and 8 poor - .81% poor.
For 802.11n: 390 good calls and 14 poor - 3.47% poor.

Any thought as to how radio type would affect call quality to that degree?
Hello experts,

I was moving the phone numbers to the cloud and as I was doing that the account manager also included an analog line that was connected to the PA system. The analog number is also now moved to the cloud and it is showing as spare. I have asked the account manager to release it back so the PA SYSTEM can work again.

Do you know if the cloud phone vendor removes that number , if the PA System will work? or do I need to do anything from my end, the PA system is cross connected with the copper line to the BIX.

Another thing I noticed is that in the call manager the extensions are associated to MAC addresses .

I just need some direction on this .
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Hi, Need Help
                           I bought these Used NEC phones DT700 32 BUTTON for my Office, we are Using NEC SV8100 PBX. All the old phones are working perfectly fine,
                           But the New Used Phones I bought is giving issues by saying SIP Server not Found. .

                           I have tried a couple of things but no luck so far.
                                 I have Hard reset the phone first to clear all the old settings.

                                 Enable the DHCP Mode,  Entered the SIP Server IP Address, and enter the SIP Extension to be used, but No luck,

                            The same error says SIP Server not found.

                          I have even created a new Ext. on Web Pro with new Port, but no progress.

                          Is there is any specific configuration needs to be done before adding DT700 to SV8100, If Yes, What should I do?
                           I bought 5 phones like this, and all of that 5 is not working.

                     Can anyone help, please?

I have 1000+ users that still use 3rd party audio conferencing with Skype for Business.  
I see that these have to be moved over to Microsoft by April 1st:  https://docs.microsoft.com/en-us/skypeforbusiness/legal-and-regulatory/end-of-integration-with-3rd-party-providers
I need to get a list of all current users and what provider they have assigned, then migrate from a list.
Referencing this:  https://docs.microsoft.com/en-us/powershell/module/skype/get-csonlinedialinconferencinguserinfo?view=skype-ps - I'm getting errors:
Get-CsOnlineDialInConferencingUserInfo -Filter {Provider -eq "InterCall"} -First 10

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Cmdlet invocation error
    + CategoryInfo          : NotSpecified: (:) [Get-CsOnlineDialInConferencingUserInfo], CmdletInvocationException
    + FullyQualifiedErrorId : Error processing cmdlet request,Microsoft.Rtc.Management.Hosted.Cbd.GetCsOnlineDialInConferencingUserInfoCmdlet
    + PSComputerName        : admin0a.online.lync.com
Or, if I run:  
Get-CsOnlineDialInConferencingUserInfo -Select ConferencingProviderOther

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- I only get a limited number of results back and there is no "resultsize" filter available.

Or if I try to gather from this command, I'm only getting around 50 results back, and there is no "resultsize" filter available.
Get-CsOnlineDialInConferencingUserInfo -Select NoFilter | select displayname, provider 

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Any help is greatly appreciated.
My customer has 11 offices distributed over 11 states, for each of those states this customer uses Skype for business Client (endpoint) to make/receive calls.
They have a main toll free number which their clients call them on.
I would like to route the caller based on their state they are calling from to the office number.

I know this can be done via FreePBX because I have already done a test for 3 numbers however in order to go through the steps of uploading the database of NPA, States and creating routes based on these numbers (state codes) I would like to know the step by step procedure to do so.

I have asked in the FreePBX forum and they have given me the how to but it's not clear to me how to do so because I am fairly new to the batches and scripts on FreePBX bash. I would appreciate any help .

I am writing down call follow and how things are supposed to work.

My Customer Toll free Number = 888-XXX-XXX
Every one of their offices has a main number = 877-XXX-XX1/2/3-12

Assuming I called from Newyork with CID 203-XXX-XXX to the Main Toll Free 888-XXX-XXX the call in this case should be routed to NewYork's office number.
I already built the CSV file which has all codes/states abbreviations but need to know how to build routes based on this.

Thank you
Hi Experts.

I am in a strange situation with a 1-888 number my organization is using , we have the same 1888 number in Canada and US. I am in Canada and when I dial this 1-888 # it is being picked up by a person and that is good . but when Someone dials the 1-888 number in TEXAS and I am informed by our staff in TEXAS that it should ring to a local number.

My question is this number 1-888 supplied by the same carrier? Can the same number be supplied by different carriers at the same time ? I know the carrier in Canada but I do not know the carrier in the US.

Please assist

We are currently evaluating option to move our voice to hosted.  We are in the process of two part project for this.  1st is migrating from VPN to MPLS.  The 2nd is to move from PBX/SIP to hosted.

Currently using Shortel and planning on gamma.

Anyone gone this route and suggest any options or caveats?

I was trying to bring up a SIP over IPSec peering on a CIsco ISR4451-X. This is to complement and already existing
SIP router (CUBE I guess they call it). Anyhow I had a call not connecting and I ran debug ccsip call. One weird thing
I noticed is that the Source IP Media address address on the external Gig 0/1 interface of the router.
The loopback is and that's what I'm expecting to see for Source IP Address (Media). What determines
what interface/address gets used for Source IP Address (Media)? Thank you.

001812: Dec 17 2018 23:26:18.025 PST: //747/E74BC54D82EC/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711ulaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 0 (tx), 0 (rx)
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 101 (tx), 101 (rx)
Source IP Address (Media):        
Source IP Port    (Media): 17876
Destn  IP Address (Media):
Destn  IP Port    (Media): 55414
Orig Destn IP Address:Port (Media): [ - ]:0
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.
If I have two SIP routes - model 2951 ISRs CUBE - and you want call manager to
failover if one of them can't complete a call - what is required? We currently have
a SIP trunk to one ISR (and the ISR has a TIP trunk to our call center). For redundancy
we want to add a second ISR/SIP Trunk. But the second should only be used in the
event that the SIP peering on the primary goes down. Advice appreciated.
How can I set my Skype default options to be that anyone can present/share files when Skype is opened? My current set up means that I have a Skype meeting and then when the person wants to share I have to change the settings and then re-commence the meeting? It's set so that only people from my organisation can share...thanks
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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?
It was ugly with Skype. Still haven't figured how how to add (or invite) external contacts to chat in Teams
Is Cisco UCM  Version 12, can I do the following:

  1. Set a user's voicemail to allow breaking out back to the main menu
  2. Monitor Hunt Group and Call Volumes in Real Time
  3. Monitor Agent Login/Logged Out StatE?
Dear Experts,

I have two FreePBX virtual machines distributed over two different data centers but both are for the same company, I am planning to use sip trunk on those two VMs and seeking to get them to work in failover/load balance mode.

I know that some sip trunk provider provide the capabilities of failover in case my primary Public IP is not responsive however I would like to extend the failover and load balance to the level of the VM as well.

Have anyone tried to do the load balance/failover method on a VM level between two datacenters ? How would VoIP traffic react in case of primary VM down? how about end points configuration ? Can I direct end points to a single FQDN where both IPs can resolve and if one of the VMs are down still the end point would get register and act like nothing is happening ?

I would appreciate any person's comment that have had an experience with such scenario.

Thank you
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
Looking for a VoIP phone system.
This client currently has a PBX black box supporting over 50+ phones. They are renting 5 lines from a local teleco.
They are building a very modern 100K square foot facility and will be moving to there in the near future. I started researching a new phone system to replace the current one.
I am not a telephone guy, so I would like to hear your opinions. First of all, I guess VoIP is the way to go, correct? I am kind of an old school and did not trust VoIP at the beginning. But after using Skype calling long distance for years, I find it has better voice quality than the landline most of the time.
I heard of 3CX, a cloud based phone system with no need to run an in-house PBX hardware or software. If the PBX is virtually on the cloud, how can we connect the 5 rented lines from the teleco to the virtual PBX?    
What solution would you recommend based on your real world experience? Reliability, voice quality, easy management.


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.