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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AWS, Speech Recognition and AWS

I have heard AWS has an excellent Speech Recognition web API, and wonder if anyone knows of a complete speech recognition app which uses this AWS API?

Apparently, the demo emails you the translation and I hear the translation is great.

Anybody got any leans on an app that can handle speech to text?

Thanks.
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Learn Ruby Fundamentals
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Learn Ruby Fundamentals

This course will introduce you to Ruby, as well as teach you about classes, methods, variables, data structures, loops, enumerable methods, and finishing touches.

Good afternoon everyone.   I was wondering if anyone had any experience using the My Fax application.   Currently one of our customer is using the service to send outbound faxes directly from their EMR application.   The majority of the fax numbers associated with the EMR application they send to are coming through without a problem.   There is one fax number (which I believe to be an analog line) when sending , the EMR application shows the fax sends and completes the send however only partial pages are coming through.   This is only happening with 1 fax line out of the rest.   According to the vendor they did state this.

"Good afternoon. Pls note that this change reflects ALL faxes sent. In our experience, this raises a few additional issues:
 
1) Additional Support Calls in to our team as to why the failed confirmations are delayed (ie could take up to 30 minutes to produce a failed confirmation)
2) If the line we are dialing is of poor quality and cuts off multiple times mid transmission, we will still try multiple times to deliver that fax meaning that they would receive partial faxes…."

Where specifically can I look to find out where the break is happening?   Is there a way to verify if the line quality is poor?  Or if not, and the fax is making it to the destination could the drop happen on the way they are processing the inbound transmission on their end?

Here is some additional information from the vendor..

I have investigated and tested the number …
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Teams VOIP not ringing on all devices.

I am currently doing a POC to test Teams VOIP and after I moved my user to teams only mode I got the calls to route thru my teams desktop app on my laptop but I am also logged into teams on a Yealink Teams Physical Phone and the Team Mobil App on my iPhone. Is there a way to get all 3 devices to ring? Or how to pick what device will ring?
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We have a Fortinet FortiWifi 30e connected to BT fibre.  There is a single software switch including all 4 ports. There is a DHCP server and scope (192.168.5.x) running on the internal interface.

I have defined VLAN_10, set the VLAN ID as 10 and created a second DHCP server and scope (192.168.10.x).

There is a single cable running from port 1 on the Fortinet to a ubiquiti 16 port POE switch.  5 yealink T46s phones are directly connected to the Ubiquiti switch and 5 laptops connected via the second port on the back of each phone.  All devices get IP addresses from the internal range i.e. 5.x

When I set the VLAN tag on a laptop to 10 it will get an IP address from the 2nd scope i.e. 10.x as expected.


BT provided the phones and will not allow us access to the configuration so I can set the VLAN tag on the phones to 10.  They have suggested that we enable LLDP-MED for VLAN_10 and this will force the phones to pickup  an address from the 2nd scope.

Any suggestions as to how this is done? (preferably in noob speak)
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Hi there,

I just upgraded the firmware on one of our PowerConnect 6248P VOIP switches to version 3.3.18.1.  But even though the switch is up and running, I can't connect to the management interface website anymore.  Every time I go to the switches management IP from a browser, the site never comes up.  Is there a quick solution to getting the site back up again?
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How do you configure a Polycom VVX 400 to stop switching the time and date on and off?

Visual: https://www.youtube.com/watch?v=X4nQjxz3e28

The Polycom phone  is linked to Skype for Business.
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We have a yealink IP SIP phone that needs to connect from the outside.  We have set up the phone and tested it internally and it's good so we moved it externally.  

In the phone, we edited the account and put in the public IP of our router (a Sonicwall NSA4600) and waited for the phone to register.  It fails to register.

I've checked the ports and I DO have the right ports open on the Sonicwall but I've also read a lot of posts about people having trouble with SW and SIP phones.  So, after reading, I have made the changes suggested on the posts I've read and still no joy.  

I'm using https://www.yougetsignal.com/ to test 5060 and it reports that it's closed.

Does anyone have any insight into how to make the SW work well with the SIP phone?

PS:  An alternate port scanning tool tells me the port is filtered.  So, I'm looking up how to turn filtering off in the sonicwall for this service.

Thanks
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QoS Expedited Forwarding (EF) value of 46

Can someone explain the Value 46 of EF  ,where did 46 come from ?

reading online I found this :
 "DSCP 46 is backward compatible with an IP Precedence value of 5 as seen in the following binary pattern: 101110 = DSCP 46 "

Not clear enough though

Any QoS  to elaborate on this ?

Thanks
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Hello,

I am a beginner with 3CX, and unfortunately I'm having issues configuring my new Grandstream GXW4104 gateway. I can get outbound calls to work, but for the life of me I can't figure out how to get the inbound calls to work properly. I've spent 2 days trying different things, tried several configuration guides, read the forums, been all over the internet, but nothing seems to work. I did have 1 successful inbound call routed to an extension (immediately after completing a re-configuration), but was unable to make a 2nd inbound call - it just rings out.

Strangely, after a standard configuration (as per the 3CX guide), with no inbound rules, inbound calls will consistently go to the operator, which is by default sent to voicemail, as I have no phone set up for that extension. If I then create an inbound rule or change any settings, no inbound calls come through. Changing the settings back to how they were also results in no inbound calls! I can only get it to work again by doing the gateway configuration from scratch!

Apart from the situation above, looking at the 3CX activity logs shows no activity when an inbound call is placed (log set to Verbose). Within the GXW4104 web interface, it shows that the line has a call coming through, yet 3CX reports nothing. I have set up the syslog server in the GXW4104, and from my untrained eye, when a call comes in, it is spitting out heaps of data, yet 3CX activity shows nothing. The gateway status always has a green light …
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I have a digital land line for my business, and I'd like to add an auto-attendant to that phone line, without changing my hardware, my existing phone number, or my telephone company.  The telephone company can convert my phones to a VOIP system, with an auto-attendant feature, but the cost is prohibitive.  I have heard about using Google Voice, and a VOIP converter device.  Would this be a reasonable solution, or is there another suggestion?
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Learn SQL Server Core 2016
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Learn SQL Server Core 2016

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I'd like to know how to be able to subnet a PepLink Balance one router for less than 50 users. More concretely I'd like to have the IP phones on a  separate subnet. How can I go about that?
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Cisco 7941 phone Registration Rejected.

Call Manager is good.
DHCP is good.
Did factory reset.
Tried SCCP, no go.
Went to SIP, no go.

Arg.
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I am attempting to configure VOIP multicast paging for a member school district. Currently, when a phone page is initiated, the green speaker light on the phone comes on and the mic red (mute) light comes on at same time.However, the page is not able to be heard on any phones. I am told this did work at one time but no one seems to be clear when it stopped working. As of yesterday, I upgraded their WAN switch and I am attempting to get the paging to work again.

Specifics.

Cisco Informacast server (Handles paging) is located on the WAN and is directly connected to a  vlan 110 port (near-end)
VOIP phones are located at the district on the far end of our WAN (far-end)

Near end switch connected to Informacast server on vlan 110:
HP 5412R , J9851A running KB.15.17.0007

Far end switch hands all vlan 110 VOIP traffic off to VLAN 72 (Phones):
Aruba 3810M, JL071A running KB.16.07.0003

Vlan 110 is configured on both switches and vlan 72 is only configured on the far end.


Obviously, the config is non-working (copied over from old switch) and has been modified in an attempt to resolve the issue but I will post for reference and advice on what to change.

near end switch vlan 110 config:
vlan 110
name "VLAN110"
untagged A3,B5-B6
tagged
A4-A9,A11-A19,A21-A22,A24,B1,B4,B7-B8,B13-B22,C1-C24,D1-D24,E1-E24,F1-F22
ip address 10.201.0.1 255.255.248.0
ip igmp
ip igmp forward A1,A3-A24,B1-B2,B4-B8,B10,B12-B22,C1-C24,D1-D24,E1-E24,F1-F22
qos priority 7
forbid B2
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What is the best Opensource chat or team servers that you can install on your premises ? Something like Matrix (Synapse) , RocketChat or Zulip?

Something that supports Chat and PBX integration.
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I am looking to roll out a FreePBX phone system for one of my clients. I have experience setting up FreePBX itself, though I'm not sure which SIP provider I should go with to host my phone services and numbers.

Do any of you have SIP providers you would recommend, or have any other tips I should keep in mind while rolling out VOIP services for business?

Thank you!
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Ive been researching refurbished Cisco Voice switches.  The pricing is compelling given the abilities within the equipment.  They could potentially solve a lot of problems.
The location where they will be used needs to be provided with a lot of uptime.  They are limited with budget.  The Cisco equipment is at or near end of life, but can be warranted still by Cisco.  the model numbers are the 3750X with full POE.  My research indicates they have static routes and more, but some of the queries I have seen shows that people have had problems with the same said static routes.  Does anyone have these, or used these in the past, or present. If so, is there any issue I need to be concerned with regarding Static routing.  I must have static rotes as there are about 8 subnets within the two campus settings that are to be ties together with MPLS.

If anyone has any Idea it would help immensely.  The issues is highly important as I must make the purchase today, but have this question.

Thank you
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I need to set up a switch to serve a VOIP phone system where clients connect to the network through the phone.
I've set up a ProSafe GS108v3 with 2 VLANs, VLAN 1 ports 2-8 untagged and VLAN 20 port 2-8 tagged (port 1 for connection to the firewall/router).
The Firewall/Router is set with QoS enabled

Is this sufficient/correct for use as intended, and is there anything else I should do to make this work correctly/optimally?

Thanks much.
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Hi, in a an recent update iPad update, I now have the apps "Voice memos" in my iPad; something I was hoping one day to happen since I use the apps a lot in my iPhone.  Is there a way to have the same functionality like apple notes that if I do a note in my iPhone it will appear in my iPad?

Thank you

just in case, the icon of the apps:
voice memos icon
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Dear All

I have been testing Microsoft teams direct routing and this has been working really well for me but I cannot find any documentation for the below scenario, would anyone be able to point me in the right direction?

We don’t want to give everyone in the company a DDI but rather have extension numbers so what I would like to do is have 1 main office number and all my users will have extension numbers off that. So all my users out going calls will be presented as my main office number, for incoming calls and external person will need to dial the main office number and then once answered they can dial the extension number.

Is this possible?
0
HTML5 and CSS3 Fundamentals
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HTML5 and CSS3 Fundamentals

Build a website from the ground up by first learning the fundamentals of HTML5 and CSS3, the two popular programming languages used to present content online. HTML deals with fonts, colors, graphics, and hyperlinks, while CSS describes how HTML elements are to be displayed.

After upgrading our Cisco Communications Manager to 12.5.1, we have SOME phones that are unable to get an IP address from the publisher or subscriber.  Additionally, with some other models we needed to power cycle the phones.  The model most affected is the 7841, but not all of them are problematic.  We have factory reset a bunch, set static ip addresses, all to no avail.  I am interested in what could be the issue with DHCP and any steps I should take to fix it.  The phones in question are typically stuck at "verify your network settings" screen or some other network message.

We have a case open with TAC, but I wanted to get real live end user suggestions.  Thank you for any assistance you can provide EE!
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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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At a customer site we have a VoIP server running. The customer complains that the some voice conversations are not going well. Even with collegues next door.

The network guys have checked the network and told that there is no problem on the network. They tell that the voice server has not enough troughput or that the network card has an issue.

is there a software that i can install on the server to monitor the lan? i want to know what the max network troughput asked is. Not a measure like speedtest.net i want to know how much is asked when calling.

The server is windows based to i need a windows software.
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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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I have someone who has VOIP service in Franklin North Carolina named Vanilla Soft.  The have an ATT hot spot because there is really not many ways to get internet out in the woods.  She works from her computer using VOIP and the cheap internet connection.  Sometimes it works great and other times it does not work well.  Is there anything I can troubleshoot to optimize her VOIP connection on her Windows computer.  She does not have many options to upgrade her internet line so we have to try to figure out how to make this situation better.  Thanks for your advice.
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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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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).