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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).

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I need to add dial in numbers for conference calls for my organization within Office 365.

We also need to make sure that whenever a user schedules a Skype meeting within Outlook 2016 (through Office 365) that the dial in numbers will appear below the "Join Skype Meeting" section of the calendar invitation (see the screenshot).

How can this be done?

JOIN-SKYPE-MEETING
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Front gate buzzer will not ring into building over VOIP.  can anyone suggest
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Hey Experts, we have a Digium Switchvox VoIP Server. This past weekend our local power company had to upgrade our facilities power. We gracefully shut down everything Friday night, power was restored yesterday afternoon. This morning we have half of our phones not working as they cannot get an IP now. Our LAN and VoIP LAN are attached to our SonicWALL NSA2600, we have 3 Cisco SG500-28-p Stacked switches. What we have found so far is that any phone connected to the Master switch will not get an IP for the phone. Each desk has 1 Ethernet drop, that goes into the phone and the workstation plugs into the phone. The workstations all work fine to phones that don't work. We have rebooted the switches for good measure and nothing changes. Hoping someone can help shed some light on what the problem is.

Here is how the config on the sonicwall looks for the interfaces
Interfaces on SonicWALL NSA2600
Here is the Stack.
SG500-28P Stack
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Hi all,

I have requested an additional IP address block from my ISP so that I can assign a public IP directly to my VOIP server. I have received and added a nat statement to my router as follows

ip nat inside source static 10.121.50.1 XXX.XXX.XXX.XXX (being one of the static ip's assigned by our ISP)

I can establish a SIP session with my server from outside however still get no audio either way. I ordered the additional IP so I could NAT everything from the external ip to the server to avoid this exact issue however it hasn't worked. To me it looks like no traffic is going back out the nat statement as the debug always shows 0 packets going out but plenty going in

*Jan 15 15:32:53.900: NAT*: s=183.171.81.177, d=58.XX.XX.X->10.121.50.1 [46336]
*Jan 15 15:32:53.960: NAT*: s=183.171.81.177, d=58.XX.XX.XX->10.121.50.1 [28621]
*Jan 15 15:32:54.208: NAT*: s=10.121.50.1->58.XX.XX.XX, d=183.171.81.177 [0]
*Jan 15 15:32:54.212: NAT*: s=10.121.50.1->58.XX.XX.XX, d=183.171.81.177 [0]

183.171.81.177 is my handphone on 4G  
58.XX.XX.XX public IP
Any help Appreciated
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I currently have 1 PRI configured on my voice gateway router.  We have had a few instances where we have had 20-21 simultaneous calls at a time, and as you know a single PRI only allows for 23 simultaneous calls.   I am looking to get another PRI from the same telco.  How does this work?  I have another T1(PRI) port on my router, which will be used to connect to the 2nd PRI, but how does it work on the Teclo side?  Do they  trunk the two PRI's together, so I can now have 46 simultaneous calls?  We are going to order another block of DID's with this new PRI as well.  So right now there are 39 numbers associated with the first T1, and I'm not sure yet how many we are going to get with the second block.    Does the telco tie these two PRI's together somehow, so both PRI's can share all the numbers?
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OK don't laugh. I have a 9 year old Cisco call manager which has run flawlessly for the last 9 years.

Recently it has developed a problem, since it has been end of life'd by Cisco they will not help me with this issue.

Here is the problem.

When a user goes to listen to their voice mail, everything works properly it will tell them they have "X" amount of messages, To listen to your messages press 1...

Once they press one again it works as normal... Saying a message from.... sent on....

Then right when it would normally play that message,

a message will play that says.

"This message contains no recording."  
Then it will go on with the normal  to save it press 2 to delete it press 3

No matter what option you select the next message played is.

"this system is temporarily  unable to complete your call, call gain later, good bye."

On the previous step If you press 2, to save the message. And go into saved messages it is there.

Since I have 90 mailboxes and get over 200 messages a day this is becoming a huge issue.

I'm hoping someone here may have enough knowledge or could at minimum refer me to someone who can help me band aid this until I can work on a replacement plan...

Thanks

Here are some version screen shots.

Show Hardware
https://www.screencast.com/t/WWcVu2wO 

Show System
https://www.screencast.com/t/rgFoIOWBb7
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Can anyone point me in the right direction on how to configure cell phones to  use with Call Manager 11.5 ?

Can this be configured for use with or without VPN ?

I believe express way needs to be configured ?
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using windows 10 and google voice (pick one browser)

how can I record phone calls
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I recently moved a small Call Center from one location to another. They contacted me late in the game to help them get a new call center setup with very little time. This left me with Comcast as the only ISP to choose from that could install services within their timetable. We decided to go with their new Gigabit package, which includes their new Gb modem. The client also chose to go with Jive for their PBX needs and subsequently ordered 20 Yealink T40P IP phones. Let the issues begin...

Comcast's modem would partially default to factory settings once a week. When I say partial, I mean things such as factory internal Gateway IP and Subnet revert back; passwords are set to default, yet the SSID's are still what we created. Comcast claims that they can ping the modem but when I test connectivity inside the modem to the internet I get 0/4 packets received. This test was run by the modem internal troubleshooting software. The Uptime clock inside the modem says 17,500+ days, which is not possible since we have reset the modem within the last 24 hours and we have only had the service for three weeks. We have factory reset the modem, replaced it with a new one, and we are still having the Yealink phones not obtain an IP address, or they have an IP address but say "No Service", even after they have previously worked. Phones continue to drop off of the network in that manner, and PC's that are using pass-through are losing connectivity as well. No matter how many factory resets on …
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We are still attempting to resolve random VOIP issues ( dropped calls, 1/2 rings then straight to voicemail, etc...) We have implemented a new Meraki MR320 switch and placed all desktops/phones (daisy chained) on this new switch.  We have this Meraki switch uplink to physical port on our firewall. There are no other switches connected to the Meraki. Physical port on firewall that Meraki is connected to is part of a bridge to another port on firewall which contains rest of our network switches. We have 2 vlans, voice and data setup on Meraki. I am seeing the messages on the Meraki log below over and over randomly during the day. Is this an issue? I have done some research on RSTP and port settings for the Meraki but still not sure I have the full grasp and if what I am seeing is actually an issue. FYI, we do not have devices being unplugged when seeing the issues. We have newer cabling to all ports.

STP Log
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QoS and roaming policies, that kind of thing.
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I am looking to install a small low cost WIRELESS VOIP phone system for my church.  Requirements are simple - VOIP phones that will operate via the Church's 802.11 wireless network.  We have 2 incoming analog telephone lines.  Will need 5-10 telephones with voicemail on each.  Would like a desktop style phone in most locations (vs. a small handheld).  Recommendations please.
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I have only one standard copper pair phone line. I very frequently use this line to play recordings of voice and music from my computer to the phone line using a Plantronics MX10 interface device. I also legally record some phone calls. These capabilities are ones that I MUST have. Lately I am having SUBSTANTIAL difficulty dealing with the "Phone Company" and employees that have little knowledge and even less desire to solve problems. I am sick of spending so much time trying to reach someone that can and will solve the problem. Yesterday I was on the phone for slightly over three hours! I got nowhere.
I know very little about voip or magic jacks and such. Is there another solution for me that will allow me to continue to leave my computer hooked up to a copper pair so that I can continue to use the phone system the way I utilize it presently? My isp suggested that I purchase a MagicJack PLUS but also was candid with me that he was not completely sure it would accomplish my goals.

(ASIDE) For those members that are perhaps wondering what is at the heart of this problem I will say that I treat people as I would like to be treated....honestly. I am being overcharged for normal phone service and refuse to be cheated every month for something I am not receiving. Getting rid of the entire companies service 100% would end my problems with Century _ink for good.
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I have a 3750G issuing DHCP to a bunch of voip phones and mostly seems to work.
I have option 66 configured
ip dhcp pool Phones
   network 192.168.200.0 255.255.255.0
   default-router 192.168.200.254
   option 66 ip 192.168.200.200

Open in new window


but when I look at the phones GUI I see this odd value
	DHCP Option Server Path	@(HH

Open in new window

I'm not even sure what question to ask other than is there a different way I'm supposed to set option 66?
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I know a little about VOIP so please bear with me on this <g>

I know of services like Vonage that use an adapter - plug the ethernet into the box and it has an rj11 for a regular phone.

Then there are services that use a 'VOIP phone' like the grandstream GXP2130.  Is there a word or phrase that distinguishes these 2 type of services?

or most any service can use both - the adapter or voip phone?

And specifically, i have a spare GXP2130 phone. I'd like to know if I can set that up to make & receive calls using google voice?

or are there other free / very cheap services? I don't care about reliability really.  if its down for hours at a time not a big deal. not looking to port a number to the service. Just have another line / experiment with this gxp2130
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We have a user who is moving to another city and we want to set up his Mitel phone so it is like he is still in the office using his same extension as before. I believe we have to setup a vpn so we can do without using teleworker feature on the phone but I am unsure. Looking for advice on how to go about setting up this user to have his phone work remotely. The only VPN we currently use is Cisco AnyConnect but just want to know what steps it would take to accomplish. Thanks in advance for the help.
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Dear All,

Thank you all for your support.I have a user that has skype issue whereby when he tries to join Skype meeting his Skype freezes & when he tries to host a Skype meeting it just hangs there no visual/audio display.

Any ideas much appreciated.
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I am trying to get Zulu UC Softphone to work with FreePBX  13.0.192.16

I have researched virtually every article I can find related to this issue, with no success.

I have the Zulu UC Softphone working on a client.  But cannot get the URL Popup to work.

I get this error on the client window when a call comes in and not sure if it's related or not.


Connection dropped by remote peer.

Any help would greatly be appreciated.


Patrick
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for a small business which conference system will you recommend?

what are the leading brands out there?
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I am using an SSG-140 Firewall at our rack in our data centre. Behind the firewall are several VMWare host servers which house a bunch of VP Servers, one of which hosts several 3CX VOIP PBXes for our customers.

I'm wanting to setup VPN tunnels for each customer so that we can do away with the SBC that 3CX demands we use if remote.

Can someone tell me how to set up a VPN connection to the firewall for our "test" customer (me!) so I can check it works? I'm using the WEB interface on the Juniper as I'm not too clued up with the CLI at this point on the unit.

The VPN tunnel should be ALWAYS UP, LAN to LAN and STATIC IP to STATIC IP.

Any help gratefully accepted.

Cheers
Chris
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i need to know area code of Uk, because need to call up to London.
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windows 10 chrome browser

Using google voice, I want to call a government worker.
Phone rings to voicemail.
This government worker does not call back if I leave voicemail

So I want to set up a way to call numerous times thoughout the day but disconnect if phone reaches voicemail
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windows 10 chrome browser


google hangouts
"your call is free" dialog box
just an extra click if I make many phone calls

is there a way to make a phone call with less google voice clicks
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windows 10 chrome browser

https://chrome.google.com/webstore/detail/google-voice-by-google/kcnhkahnjcbndmmehfkdnkjomaanaooo/related?hl=en
This extension does not have click to call

http://techextension.com/click-to-call-chrome-extension.php
many pop ups but I could not figure how to make it work with google voice
many choices of integration but google voice not included

Is there a google voice click to call chrome extension so I dont need to copy and paste phone numbers to make phone call
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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).