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IP Telephony

IP telephony (Internet Protocol telephony) is a general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). Using the Internet, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN. The challenge in IP telephony is to deliver the voice, fax, or video packets in a dependable flow to the user.

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Is Cisco UCM  Version 12, can I do the following:

  1. Set a user's voicemail to allow breaking out back to the main menu
  2. Monitor Hunt Group and Call Volumes in Real Time
  3. Monitor Agent Login/Logged Out StatE?
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Dear Experts,

I have two FreePBX virtual machines distributed over two different data centers but both are for the same company, I am planning to use sip trunk on those two VMs and seeking to get them to work in failover/load balance mode.

I know that some sip trunk provider provide the capabilities of failover in case my primary Public IP is not responsive however I would like to extend the failover and load balance to the level of the VM as well.

Have anyone tried to do the load balance/failover method on a VM level between two datacenters ? How would VoIP traffic react in case of primary VM down? how about end points configuration ? Can I direct end points to a single FQDN where both IPs can resolve and if one of the VMs are down still the end point would get register and act like nothing is happening ?

I would appreciate any person's comment that have had an experience with such scenario.

Thank you
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We have a shoretel system that is not under warranty support anymore that we have been asked to do some hopefully minimal support on as part of our day-to-day IT support for a client.

the client has 4 different locations with shoretel IP420/IP480 phones.
I am trying to move a phone from one site to another.....and I can not get the phone to show up in the list of available phones at the new site.
the phone has a static IP on the proper LAN - no VLAN.....i went over the phone menu settings with a working phone to make sure all the settings are correct on the phone.
the phone tries to download a configuration from the server - but says it failed downloading

what am I missing?  maybe licensing?
are there log files on the server side I can look at to see what failed?

as always - any help appreciated!
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we are looking for 800 phone no with api. meaning when someone call 800 phone no. we want api to be called immedately to www.myweb.com/phone no
then redirect to 310.222.2222

do you know which provider can. do it? just like something out of the box without too much coding involved
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In Skype for Business 2015, are there any APIs that I can use to query if a user has Enterprise Voice enabled?

I need to be able to differentiate users that have calling capabilities vs users that do not. Maybe being able to query if a user has a phone number associated with them would suffice too.
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Hi,

We use Mitel 5212 IP Phones. we are trying to get them to work on a custom VLAN setup on a watchguard m500 firewall. We have created the custom vlan and the ip scope which works fine. I have mimicked the DHCP options from our windows based dhcp server, however this didn't work. On the DHCP windows based DHCP server the options are:

128 Mitel TFTP xxxx.xxxx.xxxx.xxxx
129 Mitel RTC xxxx.xxxx.xxxx.xxxx
130 Mitel IP Phone Identifier MITEL IP PHONE
132 VLAN for Mitel IP Phone 0x3
133 priority for Mitel IP Phone 0x6

On the firewall dhcp scop options 9All custom)
Code       Name                                 Type            Value
128         Mitel TFTP                           IP                  xxxxxx
129         Mitel RTC                            IP                    xxxxxx
130        Mitel IP Phone Identifier  Text              MITEL IP PHONE
132        VLAN for Mitel IP Phone   Hex              3
133        Priority for Mitel IP phone Hex             6

When the phone eventually boots it gets a crazy VLAN id. Any clues as to what I am issing, or a how to guide on getting the IP phones to work?

Cheers,
Paul
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Dear Experts, I have a question related to telephony service. We are using IP PBX Grandstream UCM6510 with SIP trunking from The Provider.

So as my understanding, for example if our number is +AA 710xxxxx; I create a conference room in UCM6510 at ext 8888; then when customers want to join a conference room with us, they will call to +AA 710xxxxx, press 8888. Am I right? (AA is my country code)

But the Boss now have some customers in USA, UK,... and he wants his customers will call to USA, UK numbers, (for example: +1xxxxxxxxx; +44yyyyyyyy) respectively instead of our number (+AA 710xxxxx)  to join our conference room.

Is this feasible? Can you please suggest the solution? Many thanks!
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Dear Wizards, is there any tool for simulating Grandstream UCM6510?

just like in Network we have GNS3, EVE-NG, Packet tracer,,...Can you please suggest? Many thanks!
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I have one DC  with DHCP and DNS all in one. I am trying to connect a phone but it does not get an IP from the DHCP, Rebooted the server still getting (The DHCP service failed to see a directory server for authorization) error.

The phone (Cisco IP phone SPA 504G)  just sits on utilization network.
All other devices get IP and the lease time is set to 1 day.  It is when I try and add a new phone.
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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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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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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?
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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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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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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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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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How can I troubleshoot a Cisco IP Communicator phone stuck at "Configuring IP" during boot up? I've configured the device and added an extension and restarted the tftp server on the subscriber. But registration appears stuck. Where can I go to see logging as to what's failing?
CIPCPIC.png
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PMI ACP® Project Management
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Ok, so Windows 10 made some changes and i can no longer use the installed TAPI Driver to initiate phone calls from my access database. After some research, i am able to perform the following.

By Clicking on the start button and typing run then pressing enter the run windows opens.  I then can type this command tel:97025551212 and the tel protocol will dial the number using our Voip phones.

I also can get the computer to dial our phones by going to a command prompt and entering Start tel:97025551212

I need to be able to do this function within Microsoft Access VBA.

Any ideas anyuone.
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Hi, we have a data network here with Cisco switches that we manage. Now, there is a also a VOIP vendor who his own switches in our network. (He didnt want to use ours and VLAN everything).

Both the networks are on different subnets. Now, we noticed all of a sudden PC's started getting IP's from the phone subnet...and it's wreaking havoc internally. I tried to manually trace all 200+ cables in the office to see if someone plugged a phone device into the data network, but no luck..

How else can i troubleshoot this from say a switch level?
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I have problem with Cisco ATA 190. Incoming fax is not working. There is no issues with the Outgoing.

Call flow: Vendor==>GATEWAY==>CUCM==>Cisco ATA 190

did a debug and proper dial-peer are matching?
When I am calling externally and internally Fax beep sound can hear.
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help with setting up VLAN on a few switches for phone and data equipment.

i have a series of managed switches that are uplinked together.  I would like to set up a VLAN 100 for a dedicated router that is on port 48 of one switch. This router will listen to requests from phones that are plugged into any other random ports on the switch.  This switch is a ubiquiti unit that allows me to set port 48 to listen to vlan 100 traffic only.

The phones are set to 802.1Q with a vlan of 100.  there are other computers and servers on the switch that are on a 192.168.0.x subnet.  The server is handing out DHCP as well as the router on port 48.  The idea is to isolate the traffic for the phones to ONLY communicate with the DHCP server on port 48.  

Right now, this setting is working. However my question to you, is since the phones are all plugged into random ports 1-47 and set to vlan100  and these ports are set to listen to both default lan traffic as well as vlan100...am i simply congesting the switch with added default and vlan traffic vs setting the actual ports that the phones are plugged into to ONLY vlan 100?

Also, if i plug in another switch,, do i need to set the uplink from one switch to another switch with a vlan100 for them to comminicate or will they pass the phones traffic that is tagged 802.1Q VLAN 100 traffic to the other where the port 48 will ultimately listen and grab it? Thank you!
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We recently installed a Polycom  RealPresence Trio 8800 in one of our conference rooms.  We are currently on Office 365 with Skype For Business.  We created a generic conference room 'user account' that we have logged into the device.  The generic account has an Office 365 E3 license assigned.  However, when using Skype, we cannot access the global directory to invite other users.  Even if I log in as myself I only get my Skype Contacts but cannot get out to the Global directory.  What are we doing wrong?
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Hello,

Are cloud phones better than an on premise solution?
We currently have a Siemens 4K phone system used with a PRI.
We have about 230 voice over IP phones and about 15 digital line phones.
Most of the ip phones have a basic setup with add on module to configure additional lines.

The 15 digital lines are for our sales department, the have a ACD routing setup(similar to a hunt group). Each sales person has their prime line and 2 secondary lines. We have it configured so that that each sales person is able to answered each others lines. Each sales person has about 50 lines configured on two phones at their desk that they can answer when any line rings. We it setup this way because our president wants sales to try and answer all calls and not have them go to voicemail.

We recently started to look at cloud phones and providers like ring central, 101voice and others. Our main concern is QOS and if our setup in sales will be supported.

We are not sure if cloud phones will be more reliable and if we will have the same quality of service?

Thank You


Let me know if you need any clarification.
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IP Telephony

IP telephony (Internet Protocol telephony) is a general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). Using the Internet, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN. The challenge in IP telephony is to deliver the voice, fax, or video packets in a dependable flow to the user.