Voice Over IP

Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).  Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).

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Could someone please help me understand what needs to be done transitioning from Cisco VOIP to Avaya?  Specifically, Option 242 that needs to be configured on DHCP Voice Pool? The client claims that the phones are staying in Data Vlan (Vlan 1) and not going over to Voice Vlan (Vlan 900).

The setup is pretty straight forward. There is one building (3 closets total) and only 2 Vlans: Data and Voice.  

Vlan 1 10.12.0.0/16 Data
Vlan 900 10.13.0.0/16 Voice

The Cisco 2921 Router is configured for Voice DHCP like this:
!
ip dhcp pool VOIP
 network 10.13.0.0 255.255.0.0
 default-router 10.13.0.1
 option 150 ip 10.13.0.10
 dns-server 4.2.2.2

10.13.0.10 is the Call Manager IP.

I believe there is an Avaya server (controller) onsite that will be acting as DHCP for the new Avaya phones.

What needs to be done for the Avaya phones to obtain 10.13.x.x IP addresses and not 10.12.x.x ? I've read about Options 242 (or 176) that need to be configured but I'm not sure how to go about it. I haven't really worked with Avaya phones before.

What about LLDP? The switches are Cisco, 2960x. Switchport are configured to access vlan 1 and Voice vlan 900.

Any help would be appreciated!
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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 have a Vlan set in one Layer3 Dell 6248 switch, this vlan purpose is to connect our PBX, I have two subnets, my Native has the scope 192.168.16.0/24 and the vlan 7 has 10.11.0.0/24 now if I have my facts right I created the vlan in the switch  and gave it IP 10.11.0.5 255.255.255.0 with IP Helper Address 192.168.16.40 that in it self should be able to route to my DHCP server where I created the 10.11.0.0 scope, I think that I have everything cover so I can plug phones into the switch and they will be able to reach the 10.11.0.0 subnet to get an IP from that scope but I connect the phones that are programmed to look for vlan 7 and they cannot reach, but some of my devices that are in 192.168.16.0 scope are reporting to the  10.11.0.0 scope, can anyone help me to understand what did I do wrong ?
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DHCP.JPG
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Need to save money on Verizon Portable MiFi

I spend about $45/month for unlimited bandwidth and wonder if I can save money with pay-as you go?

I use almost none each month.

Who can I switch to and still have the Verizon cell network, or similar?

Thanks
0
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
0
Hi Experts,

We are moving to a new Mitel Cloud service and have IP485G phones.

I setup a VLAN on our switches along with a DHCP scope on a windows 2012r2 DC.


We receive the correct IP, router, dns  from the VOIP VLAN but the option .156 is not populating in order to get config files and firmware updates.

I am using the string below configServers="update.sky.shoretel.com"  

Thank You
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What is the easiest way to configure Cisco switches to use Vlans and Trunks?

- 6 Cisco switches
Fiber in
- VoIP
- PC
- IP cameras
0
We have an on-premise Skype for Business installation and require international dialin numbers for telcos.  Is there a way / service / provider / solution on how to offer international dialin numbers in combination with our on-premise Skype for Business installation?

Thanks
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Hi Experts,

I have a problem with my Cisco SPA504G VOIP handset.

The horizontal red light at the top of the phone comes on when I receive a message.

I want the red light to be off all the time, as I receive my messages by email.

I changed the Cisco SPA504G configuration, Voice / Ext 1 / Call Feature Settings / Message Waiting to "no".
That turned the light off, but it came on again when the next message was left.

Regards,
Leigh
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Hi Experts,

We are merging with another company. Our current challenge is that we will be moving users to different locations before the merger is completed. and they would bring their current  phones to  the new location and we are on 2 different systems.

The acquiring company has a Mitel cloud phone service and we have a Shortel on premise service.

The routing between both companies will be in place this week, so we should be able to route the traffic. The phones are on different VLANS, So the thought was is that we would need to add the acquiring company's subnet and VLAN ID to our layer 3 switches in order to pass the traffic out of our network to the Mitel cloud and not to our local Shoretel PBX for service

Would this work?

Thanks
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Microsoft Azure 2017
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Microsoft Azure 2017

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Hello Experts,

I want a tornado warning siren that is 2 min long. The file should be

No spaces in the filename
.WAV file
16-bit
8000Mhz
mono

Can someone direct me to a free download link?
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I am a network admin and looking at our switching infrastructure. I feel like it is not efficiently built, and it is aging. It consists of HP Procurve switches, some of them are V1910's, newer ones are 2530's. (Several were purchased to support a VoIP phone system). All switches are connected with trunks- this takes up 4 ports per trunk and only communicates at linespeed. What I think needs to happen is I need to build a new core switch infrastructure, remove the aging equipment, relegate the newer switches to edge roles for client connections, and select adequate infrastructure for the core equipment... whatever I get, I am planning to use cascades so we don't end up using half the ports to trunk and get better performance. Our network is not complex- we have an MPLS, an inter-building fiber link, 2 VoIP VLANs, and 3 subnets at this location. so we wouldn't need more than 8 VLAN's- right now everything except the fiber and the phones are on the same VLAN and separated by routers- I think that using the VLAN capabilities of the new infrastructure could replace routing equipment and optimize the network further. So my question is, assuming we are keeping the newer HP Procurves (The newest are actually Aruba's), which have GBIC's but no cascade ports, should we use Cisco's for the core switches, or stay brand-consistent with HPe/Aruba, which will become the edge switches? Any model or feature recommendations?
Thanks for the help!
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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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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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The main network is short of IPs, so we created a VLAN and moved all VoIP phones to this new VLAN. The PBX has to remain in the main network. We setup firewall rules to allow all traffic between the main network and the VLAN on both directions. All the VoIP phones on this VLAN work fine EXCEPT that
1. Paging voice do not come out from the speakers on all phones any more
2. Background music do not come out from the speakers on all phones any more.

For troubleshooting purpose, we moved a VoIP phone back to the main network, the above 2 problems disappeared right away, the phone worked as normal again.

I could be wrong, but I think we need to enable broadcast between the main network and the VLAN on the firewall. But i have no idea how to do it.
The firewall is SonicWALL TZ215

Any thoughts?

Thanks!
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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 been using a cloud based implementation of 3CX and I have 2 phones (but only use 1), a Fanvil and a Yealink.

I have a SIP trunk with 2 numbers connected to 3cx.

I am finding it a bit expensive paying for the 3cx cloud server and the sip trunk.

I wondered if there was a way to connect the SIP trunk directly to my phone without using 3cx.

Any advice would be appreciated.
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Hi
We are Migrating from a traditional internet VPN to MPLS network.  We have a shoretel PBX on premise and an ingate Siperator handling SIP calls.  We have QoS in place on switches.
Is it possible to house the SIperator at one of the sites rather than hosting in the DC?  If so how to route the traffic from Web to MPLS to site?
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

Open in new window

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

Open in new window

- 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 

Open in new window


Any help is greatly appreciated.
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We are moving into a new building, and we will have all newer Cisco switches, 2960X and 3850's for the cores.
I'm planning to have different vlans for the servers, PCs, VoIP phones, but I was thinking, since all of the different equipment need to communicate with the servers,
I will need to allow and route all the different vlans to access the servers vlan.  If that's the case, is then better to just create one flat network, everyone in one vlan, a /22 instead?
I guess I need to find some good articles on line to dig deeper into vlans, but on the surface, besides having a smaller broadcast domain, it just adds more complexity.

Any thoughts?
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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
0
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

Voice Over IP

Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).  Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).