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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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We have a Switchvox PBX server and Digium Phones on the desktop.  I am interested to learn what other people are using when employees are out on the plant floor and internal communication needs to happen.  Currently, employee has to be paged and then they have to walk to a phone and dial the extension.  It would be nice if we could use a different solution to limit the amount of time it takes to answer a call or communicate with fellow employees on the plant floor.  Thoughts?
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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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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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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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Good Morning,
I have a ShoreTel phone plugged into a Cisco 2960x and we use port security. Before I configure port-security for the end user I normally see 3 MAC addresses on the port, 1 for the PC & 2 for the phone. (1 associated to data VLAN, the other to Voice VLAN)
In this case I  do not see the phones MAC address associated to the voice VLAN.  When i configure port-security the phone shows a message of "No Service", once I remove port-security it works fine.

Has anyone seen this behavior? I am thinking it may be  a bad phone.
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Deares
 i have Cisco ASA5525 and ucm 3cx voice server i want to use external extension for makes call from anywhere  
i have one public  ip 212.122.45.66 in firewall ASA5525
and one ucm 3cx ip address local 192.168.10.10
i need to make NAT for sip port 5060 from public ip to local ip 192.168.10.10
and RTP ports 10000 to 20000 to local 192.168.10.10

anyone can help me what is the right command i have to do in cisco ASA
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Everything I've found on the Internet so far seems to indicate that there's no way to change my Skype display or username. None of the instructions seem to apply to the version of Skype I'm running – which I believe is the latest. Before I consider scrapping my current account and/or creating a new one, I just wanted to make sure I didn't have any other options.
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I call an insurance company and there is no "press 1 for english" today but there was yesterday.
And there is a long hold which wasnt there yesterday.

Todays phone call (I tried on 2 phones) goes straight to hold music.

If I dont press 1 for english, how does company know I want to speak in english?
Are buttons set up by company just for "future reasons" and the same operator always picks up?
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I've had a Skype account for many years, but haven't used it in the last couple. The other day I tried launching it, but it wouldn't work!? I noticed an icon on the desktop labeled, "SkypeSetup." So, I decided to try it. Unfortunately, my efforts have been in vain.

When I double-click the SkypeSetup icon on the desktop, I get the following:

Skype-installation-I.png
After clicking on the "I agree – next" button, I get the following dialogue:

Skype-installation-II.png
And a few more seconds of the same…

Skype-installation-III.png
Shortly thereafter, the process terminates with the following error:

Skype-installation-IV.png
After I click the OK button, I get the following dialogue:

Skype-installation-V.png
Assuming I can rectify the problem by removing the current installation of Skype, I open Control Panel, click on "Uninstall a program," and select Skype from the list of programs. When I click on "Uninstall," I get the following dialogue:

Skype-installation-VI.png
After clicking the "Yes" button, I get the following dialogue:

Skype-installation-VII.png
After a few seconds, I get the following dialogue:

Skype-installation-VIII.png
When I click the OK button, I get the following:

Skype-installation-IX.png
When I click the OK button, I get the following message:

Skype-installation-X.png
When I navigate to the ProgramData\Skype folder, I find the following…

Skype-installation-VIII-a.png
It appears as though there are two MSI files in that folder, including the one to which the uninstaller defaults. I've tried selecting each, but neither one will run. I'm not sure where to go from here.
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Hello Experts,
We are using Vicidial version 2.6-381a with Asterisk 1.4.44-VICI, I have strange issue one of my user extension 508 voice mail going into ext 801 voicemail box when using DID but when dial extension 508 it goes into right mailbox of 508, i have checked the user phone settings, DID settings to make sure DID rings and send voicemail to right extension from vicidial gui, here is Dial plan for ext 508 which is setup properly can someone please help me figure this out. Thanks

exten => 3508,1,AGI(agi://127.0.0.1:4577/call_log)
exten => 3508,n,AGI(agi-NVA_recording.agi,BOTH------Y---Y---Y)
exten => 3508,n,Wait,2
exten => 3508,n,Playback(/var/lib/asterisk/sounds/44048020)
exten => 3508,n,Wait,2
exten => 3508,n,Playback(/var/lib/asterisk/sounds/85100075)
exten => 3508,n,Wait,2
exten => 3508,n,Dial(SIP/508,15,Ttr)
exten => 3508,n,Wait,2
exten => 3508,n,Playback(/var/lib/asterisk/sounds/85100031)
exten => 3508,n,Dial(SIP/508&SIP/801&SIP/505&SIP/507&SIP/509,15,Ttr)
exten => 3508,n,Wait,2
exten => 3508,n,Playback(/var/spool/asterisk/voicemail/default/508/unavail)
exten => 3508,n,VoiceMail(508@default)
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I am pinging my voip router for my Vonage phone and I am finding lag times of over 120 ms which causes voice dropouts. How can this problem be corrected?
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I am attempting to troubleshoot our VOIP calls with packet captures and wireshark. Unfortunately, when I choose Telephony->VOIP CALLS, there is nothing displayed. I definitely testing captures during 2 test phones calls and insured protocols are enabled but still nothing. Any help would be greatly appreciated!
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Of can it work by attaching directly to CUCM as well? I think it's only with Expressway but way to check. Thank you.
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I'm adding VOIP on an existing site that uses either Cisco SRW20xx or SG300-xx switches.  I'd like you to comment on my plan for doing this:

The VOIP will be coming in from the internet on it's own connection / firewall and will be using a separate local area subnet.
It will generally be distributed through all the switches unless there's no phone at all, just computers or network devices.

There is a central LAN switch that feeds into other switches in cascade.  I will refer to this as the TOP switch here.

My plan for the downstream switches is this:
Assign VOIP VLAN 100 to all the switch ports along with the Default VLAN 1.
Trunk all the switch ports.
Tag VOIP VLAN 100.

My plan for the TOP switch is this (there being only Default VLAN 1 and VOIP VLAN 100):
Trunk all the switch ports that feed downstream switches.
Trunk any switch ports that directly feed a VOIP phone.
Leave any other ports on Default VLAN 1 in Access Mode.
Assign VOIP VLAN 100 to a single switch port that goes to the firewall.  
Make this a General Mode port joined to VOIP VLAN 100.
Manually tag this port <<< is that right?
Internet Port Setting / TaggedThe VOIP firewall won't have any VLANs set up, just a generic LAN.

Since I've never done this before, I'm a bit unclear as to whether the VOIP firewall port needs to be tagged or not BUT the port sure needs to be part of the VOIP VLAN 100 ONLY with no interVLAN routing / connection.  I want the traffic on the two VLANs to be completely …
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I have a small office with one VLAN.  There are 15 IP phones and 15 PC's that plug into a Cisco 3560G switch.  The phones and PC's are on the same VLAN.  The phones are provided via a hosted VOIP provider.  Lately we have been experience voice quality issues.  The hosted VOIP provider have asked us to prioritize the voice traffic so it has priority over all other traffic out to the WAN.  My network is very simple and looks as follows:

Cable Modem----------Cisco 2911 Router-----------Cisco 3560G switch (phones and PC's plug into this switch.)

Could someone please send me a sample config on how to accomplish this?  Thank You!
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On cisco IP phones (model number 7911) it stores some useful information about placed/received calls in the directories application- is this data stored locally on some storage within the phone, or would this be stored in a central database in a managed voip environment, if so being a cisco device can you elaborate where that information may be stored.
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I am the Office 365 administrator.

What steps need to be followed to record all Office 365 Skype for Business instant message chat conversations?
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Hello - im looking for hold music that is free to use. Anyone have a link to download one?
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Greetings,

I have a site where I have several Yealink T26P phones connected to multiple Aruba (HP) 2530 POE switches.  There is a stack of 4 switches, with the primary switch connected to a SonicWALL TZ300 router.  All the phones on the primary switch work normally.  The phones do not work on the secondary switches.  I have two VLANS (data and voice).  Basically, I have every port tagged with the voice VLAN and untagged with the data VLAN.  The only difference is that I have two ports on the primary switch that are untagged for the voice VLAN only (voice uplink from SonicWALL and phone controller).  In testing with other phone types, I've been able to plug into the POE port on any switch and have it work.  This is my first interaction with the Yealink phones, and it is remote.  As the only difference between the switches is the uplink port config, I'm guessing that I need to have a port untagged for the voice VLAN only and a different port untagged for the data VLAN only?  This would seem like a very poor design, as it would effectively require the use of up to 4 ports per switch.  So, what am I missing?  My SonicWALL is serving as the DHCP server and seems to be working fine and it actually shows the non-working phones actually get an IP.  They just don't connect.  Move them to the first switch and they immediately connect.

Thank you for any assistance that you can provide.

Jeremy
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Hello,

We are using FreePBX, and have Cisco 525G2 phones, and we added a SPA500s sidecar, how do i configure extensions on it?
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I have multiple Cisco 504g phones, that don't seem to take the attendant console changes. I have set up the first 3 extensions as a direct speed dial to other extensions on the network (the extension is set as disabled), and all 3 speed dials with flash orange, even though they are subscribed and working. I know to add c=g in the serv subscribing section of attendant console, but these phones seem to not take any changes I make. Hopefully somebody has an answer for this, as these are for a high end hotel, and I cannot have these lights constantly flashing.
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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).