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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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?
0
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 .
0
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
0
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.
0
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
0
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.
0
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?
0
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
Hello,

My company currently it's moving to a new phone system and we are stock. our DHCP it's set to IP Scope 192.168.16.xx and I created a second Scope 10.11.0.xx so it can connect via VPN tunnel with the VoIP system of our another office (we are in So. Cal and the other office in Florida) now, To my knowledge I need to create the scopes and the services on DHCP so I can setup the relay to ensure that traffic can go from the 10.11 network using the 192.168 network as gateway and at some point  create a VLAN in my switches to route.

I did all the first part until before the VLAN part, I have some problems.

1-Computers on my Scope 192.168.16.xx are registering on the 10.11.0.xx I need to know how to stop them from doing that, I need to keep them alive but without merging

2-Do I need to create a vlan to route all my VoIP traffic ? we have layer 2 switches and the router it's managed by our ISP or Do I need to setup a a new port in my firewall with that subnet routing all traffic from 10.11 to the public IP

I have a VM running server 2008 R2 as my DHCP I have 2 virtual NICS installed one running on 192.168.16.xx and the other on 10.11.0.xx
I have RRA installed with IGMP installed, and my gut tells me that I did something wrong

I have not done something like this in years so if there is anyone that can give me some guiadence I will really appreciate it.
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
Being a network administrator, among other things, I'm often asked by users to open ports in a firewall.
Usually the users don't know much about what they're asking for so they can't answer any questions - just forward what their technical people have provided.

Here is a typical example for a VOIP system:

The full network information for the VoIP system is:
Port Range (Audio): 35000-65000 UDP
Port Range (SIP): 5060 UDP, 5061 TLS
Port Range (Configuration Servers): 1024-65536 TCP source port, TCP Destination ports: 80, 443, 1443, 2443, 6716,
Port Range (Presence Servers): TCP Destination ports: 5222 and 5280.
I guess that's all well and good if you understand the context but that's where I'm not the expert.

I can set up firewall rules but, being conservative, I don't want to open incoming ports just willy-nilly in order to assure that the requestor gets what he/she wants.
If I ask them: "Are these incoming ports or outgoing ports?" they have no idea.
In some cases, I'm sure that some are outgoing.....
What I'm used to, for the most part, is that all outgoing will be allowed and all incoming will be blocked unless initiated by outgoing traffic.
Given this limited view, I would want to set up to allow incoming traffic to certain ports and leave things at that.
But, which ones?

I know this is likely a naive question.
So, in my context of understanding, how would you interpret the specification above?
And, in the details, I've never set …
0
Cisco 8851 VoIP phone.

Trying to setup Personal Directory for my contacts.

On the phone Login it asks for my UserID and PIN.

I have no idea how to determine my NetID/UserID.

Used my phone number.

No luck.


I do not have a pin, but I used the default 343842.
0
hi all,

is this possible.

network 1 - 10.3.3.0
network 2 - 192.168.50.0

they are connected via VPN. all traffic is flowing nicely apart from the phones.

the client has bought a VoIP phone system which needs to be on the same subnet, is it possible to 'trick' the 192 network to have a 1x 10.3.3.0 IP address on its network so that the phones can talk back to the phone system? And then to have the routing on the routers to move the traffic correctly.

Thanks
Gareth
0
Able to connect to a network resource okay when on Wi-Fi, but when connected to Ethernet, File Explorer hangs.

I suspect that the problem is the Ethernet connection speed, for instance, since the computer gets it Ethernet connection via a VOIP phone, which only gives the computer 10 Mbps.
The computer was replaced with a faster model.
Windows Updates were installed.
Reliability history shows the every time the File Explorer stopped responding.
My thinking is problem could be on the server itself, maybe I need to enable SMB on the system.
What else should I be looking at?
0
Putting together quotes to replace a 12 year old phone system and VOIP is an option, specifically Ring Central. They claim they have no hardware to install and don't even require QoS which I find hard to believe. They use Polycom phones that come with a high rating as well a Cisco option that comes with a high rating.

Question, has anyone actually witnessed a VOIP phone work without any additional hardware and call quality is excellent? My approach is if this is an option great for all parties involved. If it's too good to be true I prefer to put the right equipment in place to avoid any frustration after installation.

If you've used Ring Central even better, if you haven't but have success with another company please share.
0
HI Experts.

I have this policy map on most of the switches at my organization.  
Policy Map AUTOQOSPOLICY

    Class AUTOQOS_VOIP_DATA_CLASS
      set dscp ef
      police 128000 8000 exceed-action policed-dscp-transmit
    Class AUTOQOS_VOIP_SIGNAL_CLASS
      set dscp cs3
      police 32000 8000 exceed-action policed-dscp-transmit
    Class AUTOQOS_DEFAULT_CLASS
      set dscp default
      police 10000000 8000 exceed-action policed-dscp-transmit

We are now replacing the existing phones with a new cloud base phone system and they sent me these requirement for QOS and the vendor gave me this policy to use on the switches

policy-map PM-ASW-IB-User
class CM-ASW-IB-RC-Voice-RTP
set ip dscp ef
police 512000 16000 exceed-action drop
class CM-ASW-IB-RC-Video-RTP
set ip dscp af41
police 768000 8000 exceed-action policed-dscp-transmit
class CM-ASW-IB-RC-GeneralSIP
set ip dscp af31
police 32000 8000 exceed-action policed-dscp-transmit
class CM-ASW-IB-RC-Meetings-Control
set ip dscp af31
police 32000 8000 exceed-action policed-dscp-transmit
class CM-ASW-IB-RC-Other
set ip dscp af21
class CM-ASW-IB-Cust-AF13
set ip dscp af13
class CM-ASW-IB-Cust-AF12
set ip dscp af12
class CM-ASW-IB-Cust-AF11
set ip dscp af11
class class-default
set ip dscp default

Apply on the ports :

interface range Gi1/0/9-20
! no mls qos trust device cisco-phone
! no auto qos voip cisco-phone
! no mls qos trust cos
! mls qos trust dscp
! priority-queue out
! …
0
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
0
Is there software that tests noise on CAT5e and CAT6  cabling? I hired a company to test the cables and they said everything is above 1G and for the most part I believe them as I witnessed them test drops and make repairs on several damaged jacks. My concern is they didn't include a report with their invoice. Extremely disappointed as I mentioned the report was imperative for us to decide if VOIP is an option for my client.

I've insisted they put a report together and I'm aware the Fluke they used provides the one needed. While waiting on the report, and my suspicion is they forgot to turn reporting on, is there any software available that can give an indication of noise or how well VOIP will work on my clients network. Everything tested above 1G but there are still concerns I would like to eliminate. One being noise.

I've tried Solorwinds and wasn't impressed. Any other options available?
0
When our Mitel 5330e is connected directly to the wall jack it works and when the computer is connected to the wall jack data works however when we connect the p.c. to the back on the phone jack it does not work.   On the compute we get a message media disconnect when doing an ipconfig.  We have checked all cables.

Switch config
switchport access vlan 126
switchport mode access
switchport voice vlan 180
priority-queu out
mls qos trust dscp
spannin-tree portfast

Our Firewall is a Fortinet

Phone gets IP address from DCHP on 3300

Computer get IP address from DCHP firewall
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.
0
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.
0
Hi,
I would like to understand this process a bit more and the authentication flow.  Using ClearPass (similar to ISE) as a RADIUS server.

PC authenticates successfully via dot1x (EAP-TLS) when plugged into jack.  However, when plugged in via VoIP, it fails.   Discovered that the pC is not able to auth via MAB because the MAC is not in the MAC Address table.   Once added MAC to MAC Address table, PC successfully authenticates via dot1x and MAB.

What is the relation to VoIP here?  If the PC can auth successfully via dot1x(EAP-TLS) on its own, what triggers the PC to roll over to MAB and fail?
1
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
0
This is 100k sqaure feet, two storey, new manufacturing facility, on its stage of network/architect design.
We estimate that there will be 200 spots that need network access, for computers, VoIP phones, WiFi APs, smart TVs and security cameras.
In about 80 spots of the above, a VoIP phone and a PC will coexist.  These spots are the sitting spots or cubicles for engineers, managers and office staff.

The main question - should we run two network drops or one network drop in each of above 80 spots?
Option1: Two drops - One for VoIP phone, the other one for the computer.
Option2: One drop - VoIP phone and the computer will be daisy chained.

Not trying to over complicate the above main topic, we do have a few other questions as below in case you'd like to share some insights as well
1. Should we separate VoIP phones and computers into different VLANs? Why?
2. If we put VoIP phones on a separate VLAN, will the above Option2 still be doable?
3. Should we deploy CAT6A or CAT6 cables? 10G network is getting popular.
4. Should we run all cables directly from the end spots to the server room? Or, install some switches in the middle?
5. Any other thoughts? Or anything we should be aware of?

Thanks in advance!
1
It was ugly with Skype. Still haven't figured how how to add (or invite) external contacts to chat in Teams
0

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).