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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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OWASP: Threats Fundamentals
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OWASP: Threats Fundamentals

Learn the top ten threats that are present in modern web-application development and how to protect your business from them.

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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Looking for a VoIP phone system.
This client currently has a PBX black box supporting over 50+ phones. They are renting 5 lines from a local teleco.
They are building a very modern 100K square foot facility and will be moving to there in the near future. I started researching a new phone system to replace the current one.
I am not a telephone guy, so I would like to hear your opinions. First of all, I guess VoIP is the way to go, correct? I am kind of an old school and did not trust VoIP at the beginning. But after using Skype calling long distance for years, I find it has better voice quality than the landline most of the time.
I heard of 3CX, a cloud based phone system with no need to run an in-house PBX hardware or software. If the PBX is virtually on the cloud, how can we connect the 5 rented lines from the teleco to the virtual PBX?    
What solution would you recommend based on your real world experience? Reliability, voice quality, easy management.

Thanks!!
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IP address shortage on Class C network.
The company is in manufacturing business. They have Windows servers, office PCs, production PCs, network switches, internal WiFi, IP phones, machines, etc. They all consume IP addresses. Now they wanna add 40 more production PCs while there are only 20 free IP addresses.
What should be done in order to release more IPs on this network?
One thing we are considering is to create a separate network for all 20 IP phones which are used in the "sub-site". (Please see the attached diagram). We are not good at VLAN, but we can learn. Will VLAN help in this situation?  
Are there any other things we can do?
Thanks!
Jack
Map-IP-Phone.png
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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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CUCM 10.5 SIP Trunking

    I have a two site Cisco Call Manager Phone System with one server at each site (FL and California).   I have had SIP trunking up and running in Florida for about a year.   We are in the process of migrating our PRI trunks in California to SIP trunks, but we are unable to complete the RTP (Voice) connections on the calls.   Every time we attempt a call the new sip trunk which is mapped through our CA Firewall, the Call Manager Server at that site advertises the RTP IP for the server in Florida.   Since these are not mapped through the other firewall, the call fails with no audio.   I cant seem to find a way to make the secondary call manager server advertise it's own IP address for the RTP instead of using the IP of the publisher.   The calls originate from the CA server, it is just the RTP that keeps requesting to send to the wrong server.   Any help on  how to force the subscriber to advertise it's own IP or how to change it would be greatly appreciated.   At wits end on this one.
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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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PMI ACP® Project Management
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PMI ACP® Project Management

Prepare for the PMI Agile Certified Practitioner (PMI-ACP)® exam, which formally recognizes your knowledge of agile principles and your skill with agile techniques.

We have a Polycom VVX D601 IP phone AND we have a VVX D60 cordless phone that is supposed to register with the base phone.  It's not working.

We recently brought in some new fiber to the dry cleaners and with the fiber, our provider supplied IP phones (Polycom).  The dry cleaners need a cordless phone so the girls can walk around looking for clothes while talking to customers, so, we purchased a VVX D601 base and the VX D60 cordless phone that is supposed to work with the D601.

The cordless base will not pull an IP on the network.  We have plugged in the D60 to the router and it will not DHCP.  If we take the phone to our other office, an office with IP phones and a Xircom PBX, it pulls an IP.  If we take the D60 cordless to other networks that do not have phones, it will not pull an IP.  

Now, you're going to ask why we don't plug it into the switch with our other IP phones and the reason is our provider brought in a Juniper switch for the phones and they assign IPs on a static basis.  If we plug the D60 cordless into the Juniper switch provided by our phone provider, it of course will not pull and IP and our phone provider said they will not turn pan DHCP for us.  

So, my question is, since the base pulls an IP on network 1, which has IP phones and a PBX, but the cordless base will NOT pull an IP on any other network, is that because it's a phone and not a PC?

That's my guess.  The cordless D60 pulls an IP when plugged into a network with an IP phone PBX, but…
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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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can I block all voip calls
I dont want to block one phone number at a time

I dont want calls from voip phone numbers because it is usually a scam
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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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Hi guys,

I just got a new Helpsdesk user. I want to add his extension to the existing ring group in SV9100 nec phone systems.
Could some one help me with this, as I am very new to phone systems
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Hi

Does anyone know what the state is with the Shoretel (mitel) director licences?  Particularly with extensions?  We need to add more and cannot find out much info about them.
Awaiting comms co to update me.
Same question regarding the phones.  I believe 212k are EoL.  We need the line keys so what options do we have?

Thanks
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NEC SL2100 phone system started behaving poorly a week ago Sunday.  Rebooted all network equipment, and nothing changed.  Reboot the phone system and everything worked for about 20 minutes.  Same cycle of events, about same time frame.  Next checked all network equipment for firmware upgrades.  Wireless bridge Engenius EnStationAC went from 2.x to 3.1.2.11 on both sides successfully.  Problem returned yet again, but now a delay in getting dial tone of about 3-10 seconds.  If phone hangs up and connects immediately, no delay happens.  A monitoring computer is the only non-phone device on the entire voice network.  It shows random packet loss to phones.  The phone problem was originally they begain restarting at random.  They are not all happening at one time.  The delay never happens to VOIP phone on the system side of wireless bridge.  Engenius has yet to comment on the issue.  One phone (Only one phone) peforms IGMP actions accoring to Wire Shark.  Removing that specific phone solved the problem for almost 2 hours.  Yet it eventually returned.  All the time I have brought another laptop in for testing and moving it to both sides of the bridge.  If I am on the system side, I perform ping tests to the phones on the far side of bridge.  If I am testing the phone system resources, I am on the opposite side of bridge.  I perform around 1,500 packets to 3 devices simultaneously each time I test.  No latency or packet loss ever.  I have introduced a new PoE switch on the system …
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Dear Experts, is this diagram correct?

voip.PNG
We have 500 users, intend to use both VoIP service and Analog Tel service for backup. THere are 17 analog numbers to keep, so we choose Gateway GSM1024; for VoIP we use Grandstream UCM6510. We have several questions, can you please suggest?

- Can we use both UCM6510 and GSM1024 at the same time?
- Can we configure so that 1 department will use Analog line (GSM1024); other departments will use VoIP (UCM6510)?
- If one of these 2 fails , can the IP phone automatically change to the other, so that telephone service will NOT be interrupted?  
- Where can we find the reference link and installation manual for these devices?

Many thanks!
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3cx.. moved VM from one host to another and set static MAC. still no ext and cannot create TCP connection to activation.3cx.com
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Bootstrap 4: Exploring New Features
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Bootstrap 4: Exploring New Features

Learn how to use and navigate the new features included in Bootstrap 4, the most popular HTML, CSS, and JavaScript framework for developing responsive, mobile-first websites.

Is the Jabra Pro 9450 Duo Stereo headset fully compatible with VOIP services like WebEx?
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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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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!
0
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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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.