[Product update] Infrastructure Analysis Tool is now available with Business Accounts.Learn More

x

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).

Share tech news, updates, or what's on your mind.

Sign up to Post

What is the difference between a Cisco phone "factory reset" and a "wipe" of the phone.

The code to begin the "factory reset" reset is: While the phone is powering up, and before the Speaker button flashes on and off, press and hold #. ...
Release # and press 123456789*0#

The code for the wipe is:3491672850*#

What is the difference?

Also, I have a Cisco 7960 phone that will not clear IP addresses and other settings for either process.

Any ideas how do completely clear the Cisco 7960?
0
Angular Fundamentals
LVL 12
Angular Fundamentals

Learn the fundamentals of Angular 2, a JavaScript framework for developing dynamic single page applications.

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
0
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 …
0
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!
0
3cx.. moved VM from one host to another and set static MAC. still no ext and cannot create TCP connection to activation.3cx.com
0
I'm looking to implement QOS as we have some VOIP quality issues. We have Catalyst 2960-X access switches, and Nexus 9k core switches. Looking through the 2960-x Auto-QOS configuration guide, it seems too easy to be true. There's literally a few commands to run within that guide. Can it be this simple? It can't, right?

On interfaces connecting to VOIP phones:
Switch# configure terminal
Switch(config)# interface gigabitethernet x/x/x
Switch(config-if)# auto qos trust dscp
Switch(config-if)# exit

On trunk / uplink interfaces connecting to other switches:
Switch(config)# interface gigabitethernet x/x/x
Switch(config-if)# auto qos trust
Switch(config-if)# end

*Note* I'm using dscp instead of cisco-phone since they're Avaya phones and not Cisco, assuming they will be using DSCP 46 for signaling and audio.
0
Are there any tools other than Microsoft Call Quality Dashboard and Skype Analytics for measuring Skype call quality? Audio? Conerence calls? Video? We are mostly a Cisco network. There are shortcomings in the MCQ dashboard that have not been addressed since we adopted it.
0
Hi,

I have been trying to work out why my NEC SV9100 isn't displaying the name (displays number only) for a caller that's in the system address book (ABB).

For example, there is an entry in the ABB for: Joe Bloggs  123456789 but if Joe Blogs rings from 123456789 it shows that number on the screen but not Joe Blogs or both name and number.

Can anyone advise on how to fix this?  Handsets are DT800's.

Cheers.
0
Good day,

I am trying to trace and fix a RTP problem on our Asterisk PBX (13.21), to remote SIP carrier(s). Our PBX setup is like following:

Webphone (with STUN) -> PBX (No NAT) -> Intermediary Carriers -> Carriers (Tier1, Tier2)


Calls are working most of the time, however, sometimes some calls are reported to blank or dead calls when it reaches to the web phone. Investigating these calls through network, 5060 connection seems fine like following:

sdp-5060.png
Checking RTP messages, I can clearly see that the RTP coming from carrier stops at second 56 and no other media is received from carrier after that point, however PBX still tries to send the media to carrier after the answer is received.

rtp-sample.png
The above screenshot is when the RTP from carrier stops and RTP from PBX starts.

The puzzling thing is that this happens randomly, and intermediary carrier states that this happens on the tier 1 carriers as well and shoots the ball to my side. Unfortunately trying another intermediary carrier, I still face those blank call issues, but of course there is no guarantee that these two independent carriers are not using the same tier 1 or 2 carriers. It is also possible that, we just missing a small detail for this to happen.

More debug information:

Peer entry:
host=XXX.XXX.XXX.XXX
type=peer
disallow=all
allow=ulaw
allow=alaw
allow=gsm
dtmfmode=rfc2833
context=default
qualify=no
insecure=port
nat=auto_force_rport      ; 

Open in new window

0
We have installed a PBX on AWS and connected it to our on-prem Router via VPN.

My on-prem router is connected to the SIP provider via a physical connection with another on-prem MUX device (device given by sip provider).

All connections are working fine, EXCEPT, my SIP provider has a condition that all connections to their server must originate from a specific IP that they have assigned to us.

Since AWS machine is connected via VPN, all calls from PBX are picking up the IP of the AWS machine as "source IP".

For resolving this, i need to replace / masquerade / NAT / change the IPs of all connections from AWS machine's IP to SIP provider's assigned IP. Someone suggested i need NAT loopback/reflection for this. Someone also suggested packet forwarding. someone suggest IP masquerading.

Please guide how can this be done?

Regards.
Network-Diagram--1-.jpg
0
Powerful Yet Easy-to-Use Network Monitoring
Powerful Yet Easy-to-Use Network Monitoring

Identify excessive bandwidth utilization or unexpected application traffic with SolarWinds Bandwidth Analyzer Pack.

Hello,
I have about 150 CISCO 7960 phones and 50 CISCO 7942 PHONES, I AM SEEKING HELP GETTING THE 7942 PHONES UP AND RUNNING ON MY FREEPBX.... MY SERVER IS IN THE CLOUD... ANY HELP WOULD BE GREAT AND ILL PAY A FAIR PRICE TO MAKE THIS HAPPEN

RANDY
0
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?
0
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.
0
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?
0
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!
0
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.
0
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
0
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?
0
Become a Certified Penetration Testing Engineer
LVL 12
Become a Certified Penetration Testing Engineer

This CPTE Certified Penetration Testing Engineer course covers everything you need to know about becoming a Certified Penetration Testing Engineer. Career Path: Professional roles include Ethical Hackers, Security Consultants, System Administrators, and Chief Security Officers.

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
0
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.
0
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?
0
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.
0
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?
0

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).