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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In a small company that has changed recently to VoIP phones, they used to have a separate phone line to handle incoming FAX.  When they recently moved to VoIP, incoming faxes are going to a server controlled by the ISP who setup the VoIP system.  In discussion with the ISP, it was determined that certain individuals, (employees of the ISP) could see the company faxes which is not acceptable to the company.  A new Windows internal server is in the plans for the company.  Can this internal server have the ability to receive these confidential communications versus the server controlled by the ISP?  If so, could someone describe the steps and if it can be done with Windows Server?
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I want a cheap 800 number service. 3 choices with prompts. All I want is voip. There is no call center. No forwarding to cell phones. Maybe just used for voicemail. A free 1 month trial and an expensive bill is expensive.
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In Skype for Business Server 2015, Is there a way to disable clients from retrieving location information from the LIS/Secondary LIS every time the client registers with the server? I want to prevent the client from performing a HELD request to get location information when it first registers to the server. Is this possible with the on-prem server?
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Lately, our staff complain about receiving multiple spam calls daily from what appear to be 'spoofed' local numbers. It started about two weeks ago and it looks like somehow direct numbers of staff got in the hands of wrong people.

Is there something that can be done about this?


Thank you!
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Is it possible to disable the voice feature of the exchange's automated unified communications operator, without removing the transfer configuration to marked extensions directly?

For example, if a customer dials the IVR, "Thank you for calling PCH, if you know the extension number, mark it now, otherwise the menu is the following, to call sales, dial 1, support 1, administration 3"

When I disable the aforementioned option, the part of "If you know the extension number, check it now" does someone know how to avoid this problem?


I would appreciate your support.

Greetings.
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Hello and Good Morning Everyone,

          I recently added another phone line to my AT&T setup.  The hardware hookups are as follows:  The telephone/copier/fax machine has a direct telephone line connection to the back of the AT&T Gateway at the port labeled Phone Lines 1 & 2.  Then, I have another telephone line running from the Ext port of the  telephone/copier/fax machine  to the chordless phone.  That said, an AT&T agent explained to me that the  telephone/copier/fax machine with the direct connection to the Gateway will get the new activated number and the other phone, chordless one, will retain the old VoIP number.   Given this information, could someone explain what is meant by VoIP or Voice Over IP?  I did not want to bog down my AT&T agent with too many questions.  So, I decided to submit this one for review here.

            Any shared thoughts and explanations in simple terms with respect to the definition and function of VoIP will be greatly appreciated.

            Thank you

            George
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Hi,

I have a problem to establish call session between two sites over gre tunnel ipsec. The tunnel is up but I am Unable to set a call. I think the problem is Nat but I don't know how to fix it.  It's seems like the traffic were blocked in the beginning of the tunnel.

You can see the configuration files in attached.

 

Best Regards,

 

Aristide
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I'm trying to figure out if Gotomeeting can have better audio quality than SkypefBusiness. I'm talking about voice calls, where someone calls in for a meeting over the phone. Could there be a difference considering that they are both conference bridges? Can they offer better audio quality somehow?


Thank you!
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Difference between VOIP and SIP.  Could someone give me a practical description of what the main differences between VOIP and SIP are?

thanks,

Josh
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This is about the switch infrastructure using Cisco switches. Currently, there is only using one Cisco WS-2960x-48 POE switch. We also using Cisco UCS 500 series for the VOIP. We are using vlan 101 for data, and 102 for voice. Please see the attached cisco switch configuration.

Now, we intend to buy one new Cisco Meraki MS120-24 ports switch, and join this switch into the switch infrastructure. We also intend to add-in 2 more VLANs for our new VMware virtualization management and backup segments. This is a new 2-hosts virtualization (vmware), with 2 network ports to form a trunk carrying existing vlan 101 (data), management (vlan 121), and backup (vlan 122) from each host. How should I update in my existing POE switch and also the new Meraki switch? Can I make all the 3 vlans - 101, 121, and 122 routable but only allow selective ip to access. For example, only allow 192.x.x.25 to access all vlan 121 & 122 only, but not the other way round.

Thanks in advance.
Cisco-2960-48-POE-Switch.txt
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Shoretel and switch STP on/off?

Hi

Looking at replacing our switches from procurve to Aruba.   Changing the method from daisy chained via ports to a stacked method using same models.  

Im unsure if we need to have STP disabled for shoretel to function?  If this is the case we cannot stack, which i find odd.

Thanks
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How do you do!
My problem about algorithm, I don't have idea with resolving this situation.
I have two server:
1. First is ESXi on HP ProLiant G6 (rack based) - I'am creating on this server Virtual Machines for management office computers and have second VM for PBX (it's FreePBX with SCCP module for management and creating extensions for Cisco IP phone 7942G).
2. Second is simple PC keys - I use this computer for FXO PCI module with FreePBX server software. I connected city analog RJ11 lines to FXO PCI.
My problem is that - how do I receive calls from the second server (with FXO) on Cisco IP Phone (which is connected to the first server using SCCP)?
I can connecting two FreePBX between themselves with trunking. But it's working only with SIP protocol. Because, FXO lines come with SIP, I can receive calls with softphone.
Thanks to everyone for replying.
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Ok, this is the last part to getting our site paging worked out.
I need to know how I can get our ShoreTel phones speakerphones to work with Valcom speakers when a page is sent out.
We have a ShoreTel system 14.2 director and Valcom v-2003A paging controller.  We need our system to page all phones and all Valcom speakers at the same time.
Can this be accomplished?
If this scenario cannot be accomplished with our current equipment then what would I need to replace/upgrade?
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Hello everyone,
We have a ShoreTel VoIP phone system and we would like to use it to page different zones from the ShoreTel desk sets.
We have a 3 zone Valcom Page Control Unit.
Our paging goal is this:
  1. Page outside only
  2. Page inside only
  3. Page outside and inside at the same time
Is there a solution to accomplish these scenarios?
Do we replace our current Valcom 3 zone paging control unit with a new paging control unit capable of accomplishing the 3 scenarios?
Thanks
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Hi all, we need to have call recording for our VOIP system, does any know of a 3rd party vendor who offers this? Our current vendor and we are not switching anytime soon.
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I looking for suggestions for our on-call after hours clinicians.

We currently have a business need where a rotation of 10 users needs to be contacted on a rotating but changing schedule. We need to be able to route call the users personal, business and home phones. Additionally, we'd like to be able to text or message users if necessary.

What program or application are you guys using for on-call or after hours support?
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What options exist for porting a Google Voice phone number over to other services?

A couple of users who I support are interested in doing this and want to find out what other services they can port (transfer) their existing Google voice phone numbers too.
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I have installed Google voice on my Sprint PCS iPhone X.

Now how can I make my Google Voice number replace my Sprint PCS iPhone X caller ID number so that everytime that I call someone my Google Voice (and not my regular Sprint PCS phone number) will display on all outgoing calls?

Where do I go to enable this setting?

I received the email below on June 1, 2018 and I'm not sure how this affects my Sprint PCS Google voice service:

We are contacting you to let you know that per previous communication, due to upcoming upgrades to Sprint’s network, Sprint will no longer be supporting the Google Voice with Sprint integration. Effective today your Google Voice integration with your Sprint phone number (xxx) xxx-xxxx has been disabled by Sprint.

Effective today:
All outgoing calls (including international calls) and texts will be made through Sprint at Sprint’s calling and texting rates, if applicable.
All new messages, calls, and voicemails sent from your Sprint phone will not be stored in Google Voice. You will still be able to see your messages, voicemail, and call history from before June 1, 2018 in Google Voice on your Sprint device. You can also export this data from your Google Voice account at takeout.google.com.
You won’t be able to use Google Voice-enabled capabilities such as call forwarding, voicemail transcription, spam detection, and other Google Voice features. These capabilities can be enabled from your Sprint device. Click here for more information …
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In Skype for Business Server, we have 3-digit extension dialing set up. This is working just fine. The issue is that there are certain 3-digit numbers (not the least of which is 911) that we need to have go outside the local system. I'm having a difficult time getting this to work. I don't know where the issue lies, but I can't get the system to dial out to save my life. Here's what I've done:

1) Added a dialing rule for 911 and moved it to the top of the list. I have it translating to 411 for now (so as not to bombard 911 with test calls - once it works, I will make one test to 911 dispatch and ask them to confirm the caller ID and location).

Dialing Rules
2) Configured the route to match 411 and 911 (to send to the external trunk)

Voice Route
3) I even ran a Voice Routing Test Case to be sure that the result is what I expected. It is.

Voice Routing Test
However, when I dial 911 (and it gets translated to 411), it always rings our internal receptionist response group, and I can't figure out why.

Can anyone help me figure out what else I need to change to make this work?
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I have a remote extension (yealink handset) that is dropping outbound calls after 35 seconds. I have diagnosed through logging that the asterisk server is not receiving the proper ACK, and the reason for that is that the asterisk server is looking for the reply from 192.168.1.4, which is the local subnet of the remote phone. Obviously the asterisk server will never receive a reply from that address. What changes do I need to make so that the asterisk server is looking for replies from the WAN IP of the remote network? NAT is setup properly on both ends. For reference, this is CompletePBX from Xorcom that we're dealing with and a copy of the asterisk log error is below.

[2018-08-16 09:27:08] WARNING[5720] chan_sip.c: Retransmission timeout reached on transmission 405286438@192.168.1.4 for seqno 2 (Critical Response)
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I am running the newest version of Skype on my Windows 10 PC.  I need to contact someone (I have their Skype name), but when I try to connect I get the following message:

Skype Error
I don't use Skype often, and tried to find a setting or other evidence of the problem, but can't.  Can you help?

Thanks,

Phil
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We are transitioning to Skype for Business from an older product called Spark.  With Spark it is fairly easy for each user to populate their own
groups with contacts that are saved and used to IM thru skype.  How can we populate groups without adding them one user at a time. We have some fairly large
groups
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I am converting to SIP at a company site.  I set up the fortigate 60d identical to another one of our company sites where SIP is already implemented.

The SIP Cloud provider tells me that the firewall is not "passing confirmation packets  the phone system (PBX) sends out when it detects an incoming call",
and this is causing their system to transfer the call to our failover number.

I have opened up all requested ports (TCP/UDP etc), configurd policy,and QOS to high priority- everything they suggested, and as I mentioned earlier, it is configured exactly like our other site's fw.

I suspect it is a configuration with one of their servers.  In any case, Logging for all events is enabled in the firewall policy.  How can I tell if the firewall is blocking/not passing back the confirmation packets they SIp provider mentions?
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We have been using Skype for Business (formerly Lync) for IM and video conferencing but are thinking about using Skype PSTN as a replacement for our current PBX.

We are in the process of organising a trial for some users to pilot Skype for Business as their only phone option.

One of the things we need to do is gather some data once their trial has finished and I am looking for help identifying test criteria for the new system.

If you have any thoughts about the sort of questions we should be asking users to assess the suitability of Skype for Business, then please let me know.

Thoughts so far:-

Equipment
Which microphone device do you normally use for S4B phone calls?
Which amplification device do you normally use for S4B phone calls?
Frequency of use
In a typical day, how likely are you to make a phonecall using S4B
In a typical day, how likely are you to receive a phonecall using S4B
Sound Quality
How does the sound quality compare to landline
How does it compare to mobile

Any questions / comments appreciated.


Jon
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I have a client with allworx phone.  The moron didn't asked what type of speed we are running so we went from 1gb connection to 100mb connections because they connected the PC to the phones. I'm not sure if I have enough IP jacks to move the phone to a separate connection.  I was told there are splitters that will allow 1 gig for the PC.  Does anyone know about what the splitter is and where  can purchase?
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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).