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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.

Skype for business 2015 Yealink T48S | Trusted Certificate
I need assistance with phones we recently purchased, all T48S handsets with Skype for business firmware.
 
I have tried all 3 latest available firmware and stuck with this version as it offered a simpler login screen for users.
firmware
Scenario:
Phones register correctly for as long as the the trusted certificate is not present.
Periodically the handsets will populate with a CA certificate on line 1 even though everything is set to disabled below and then the users are unable to sign into the phones.

securitypage

i did some googling and found this command but its only relevant for SKype for Business online
https://docs.microsoft.com/en-us/powershell/module/skype/set-csipphonepolicy?view=skype-ps

What is causing the phones to download the internal root domain CA certificate?
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Does anyone know if there is a good soft phone or virtual phones for users to use in place of handsets?

Just wondering what people are using instead of handsets as alternatives.
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Hi there,

I'm looking for a solution to make a 3 way voice call between 3 sites and hang 5 lines off each of those sites.

Two of the sites have 1 PSTN line only and one site has two PSTN lines, and each location would like 5 POTS/physical handsets to be able to use at any given time (not cell phones).

Site A - 1 PSTN
Site B - 1 PSTN
Site C - 2 PSTN

Each location also has IP connectivity between the 3 sites but it is setup like a WAN so is a closed network and there is no internet breakout. Therefore no hosted solutions can be used.

So far this says simple PaBX either traditional or IP based.

Each site can call out which is fine but there maybe times that each of the sites need to be on the same call together and have more than one handset from each site join a call.

The bit that I'm not clear on is if each of the sites want to have a 3 way call between them and have each of the 5 local lines connect into the same call. Would the easiest way to do this be simply have a PaBX system on each of the 3 sites and have two off the sites (A and B) with 1 PaBx dial in to the site with the 2 PSTN sites (Site C). Then have the PaBX on site C merge the calls?

This seems a little clunky, is there a better way to do this? Is there an VOIP/SIP solution that would work better?

Trying to keep this uncomplicated and keep margin for error to a minimum so simple PaBX solution was my initial thought. If this is the best solution, does any one have any …
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What is the command to trace a sip call inbound or outbound on a Cisco 4431 running CUBE if I want to check that the carrier is sending me numbers correctly?
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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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I'm trying to add a VM to a user in Exchange and I keep on getting this error.  We use 3CX as our VoIP system, in a VM, and Exchange is where the VM is stored.

Starting yesterday, when I try to add a VM to a user, I keep on getting this error. NO changes to my 3cx server or Exchange, so this is odd.

I checked my event logs app/system, and didn't find anything that would point to an issue.  
Also did a wireshark capture for 10 seconds on my exchange during the time I received the error, and still, I didn't see any issues in the wireshark that could be a problem.  
I looked on my 3cx system, in the logs, and nothing shows up there as an issue.

I don't think this would have anything to do with it, but my manager disabled TLS 1.0 on our exchange server to make it more secure, so I doubt that would have anything to do with it?

Not sure what else to do, any thoughts?
exchange VM
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Does anyone know how to unlock a Shortel desktop phones (model 210 for example) to work with Ring Central?
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Hello,

I am wondering if anyone knows what cordless units are compatible with the NEC SV8100 phone system.

I need to replace a couple of cordless handsets but cannot find any, even used. There are plenty of corded DTL-12D-1 BK TEL DT300 units available but I require cordless. Ebay has some base sets but no accompanying hand unit. I'd like to get a couple of more years out of the system which is still working fine.

Thanks!
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Dear Experts, we got this requirement for PABX configuration in a hotel/resort:

ext.png
A master extension will be represented for each villa, then several sub-extensions will represented for blocks in villa. Any incoming/outgoing call will be directed via master extension. Sub-extensions will be used to calling inside a villa (between blocks) only, when calling outside, master extension shall be used.

For example the screenshot is showing Room 202 and its 6 blocks. It will have 1 master ext (8202) and 6 sub-extensions for 6 blocks.

So can NEC and Alcaltel meet this requirement? And what are the things we need to consider? Many thanks!
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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
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I am looking for a way to check the trunk and the DID charged  for the 3cx which is currently setup and working properly.
How can check these information?
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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 # on  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.

https://support.office.com/en-us/article/change-my-phone-number-for-skype-for-business-20e03cc1-c023-4e5d-bafd-064ddb59ed5e?ui=en-US&rs=en-US&ad=US


Thoughts??
skb.PNG
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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,
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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?
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Hi,
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?

Thanks
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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!

/Mats
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Hi

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
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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?
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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?

Thanks
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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.
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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
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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
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Hi

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?

Thanks
1
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 136.10.23.97 address on the external Gig 0/1 interface of the router.
The loopback is 136.10.23.98 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): 136.10.23.97        
Source IP Port    (Media): 17876
Destn  IP Address (Media): 216.25.35.21
Destn  IP Port    (Media): 55414
Orig Destn IP Address:Port (Media): [ - ]:0
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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.

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