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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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Is there software that tests noise on CAT5e and CAT6  cabling? I hired a company to test the cables and they said everything is above 1G and for the most part I believe them as I witnessed them test drops and make repairs on several damaged jacks. My concern is they didn't include a report with their invoice. Extremely disappointed as I mentioned the report was imperative for us to decide if VOIP is an option for my client.

I've insisted they put a report together and I'm aware the Fluke they used provides the one needed. While waiting on the report, and my suspicion is they forgot to turn reporting on, is there any software available that can give an indication of noise or how well VOIP will work on my clients network. Everything tested above 1G but there are still concerns I would like to eliminate. One being noise.

I've tried Solorwinds and wasn't impressed. Any other options available?
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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.

Please tell me I'm wrong:  When using S4B to call a business that has a telephone auto-attendant, our S4B dialpad works just fine.  However, if I and an employee call a business together in a S4B call, the dialpad buttons do not work.  MS tends to suggest that this is a known bug.  We're about to agree and leave it at that... and leave S4B.

But really?  What an obvious thing to need to do.  We REALLY need to do this to train our employees on calling clients, etc.

MS seems to be moving from S4B to Teams.  Teams seems to be entirely geared toward pre-scheduled meetings where all attendees have agreed to join.  This is not our need AT ALL.  We need on-the-fly ability to add a voice call to an existing voice call AND be able to punch a dialpad for auto-attendants.

Therefore, we're looking for economical (5 users or less) solutions for our VOIP needs.  MS Office 365 E3 (which we'll keep) runs us $20/month, but in addition, for Skype PTSN dialing we also need $12/month/user for Domestic Calling Plan and $8/month/user for Phone System.  Therefore, our phone system needs are $20/month/user.

Can someone recommend some economical VOIP solutions? Thank you, yes we've looked - but that process is EXTREMELY unproductive (i.e. false/misleading claims on websites, feature listings are incomplete, etc., etc.)

Thank you!
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Hi Experts,

I am able to access the call manager in our organization, I have a phone device and I can see it under Device --> Phone but I want to know how an anolog phone with DID phone number  will connect to call manager using internal extension usually using the last 4 digits as internal ext,

If the product Type Tye says : Analog Phone , does that mean it is a analog phone.
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If I have two SIP routes - model 2951 ISRs CUBE - and you want call manager to
failover if one of them can't complete a call - what is required? We currently have
a SIP trunk to one ISR (and the ISR has a TIP trunk to our call center). For redundancy
we want to add a second ISR/SIP Trunk. But the second should only be used in the
event that the SIP peering on the primary goes down. Advice appreciated.
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Im wanting to host simple PABX for multiple custmers and not sure which vendor to go for
we will host about 40 PABX's each with around 5 phones attached
requirements
support failover we will have instance in 2 separate DataCentres's for redundany/failover
needs to have single SIP trunk to host for all the calling voice channels
need to be low cost around 1$-2$ per extension and scalable as well
meed to be simple to provision and reliable
will be using yealink phones

any recommendation would be great
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One of the Shoretel Server Services showing RED. ShorewareCDRMigration-UPG
Does anyone knows about this service? It seems that server working fine. Thank you!
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How can I set my Skype default options to be that anyone can present/share files when Skype is opened? My current set up means that I have a Skype meeting and then when the person wants to share I have to change the settings and then re-commence the meeting? It's set so that only people from my organisation can share...thanks
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I have some new VoIP phones and for some reason they will not configure on my clients network, when i took them home they work perfectly. I tried Wiresharking on a hub to capture the traffic, however i am at a loss as to what it means of what is causing the issue. The DNS is our Win2012R2 server and this then forwards on to the public Google servers.
wireshark-capture.png
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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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I have to participate in an interview. I rather not but I have to do what I have to do.
Does anyone have any question on (hard and easy)
1.LAN
2.WAN
3.VoIP
4.file servers
5.Email servers
6.patch management
7.Support high-volume group printers, Xerox printers
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Python 3 Fundamentals
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Python 3 Fundamentals

This course will teach participants about installing and configuring Python, syntax, importing, statements, types, strings, booleans, files, lists, tuples, comprehensions, functions, and classes.

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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VG248 needs to be replaced. Anyone can tell basic step or send a link? Thank you
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I'm tearing my hair out on this one!

I've got a client who is having a VOIP issue (ongoing for several months).  Here are the facts:

1. Very small client: 4 users on a local domain, Windows PCs and a Server 2008R2.
2. VOIP phone system using same router as computer network.
3. Two phones work solidly.  Two are intermittent.  Tried varieties of plugging in phones, including directly into LAN ports on router instead of the network switch.
4. Intermittent symptoms are: 50% of the time a caller calls the number and gets either a fast busy signal, or "your call cannot be completed as dialed".  Sometimes the caller goes directly to voicemail even though the phone is not in use.
5. One of the phones has been replaced, as an experiment. Used to be a SNOM 320, now an Aastra model.
6. On the working phones, unplugging the phone on the network results in the caller getting voicemail.
7. On the intermittent phones, unplugging the phone typically results in the caller getting a fast busy signal (even at times when the phone seems to be working).
8. At a time when a phone was not working, I ran a continuous ping test to the phone from a computer on the LAN, and it never dropped out once.
9. Before the problems occurred, they enjoyed excellent service on this LAN hardware for some number of years.

The VOIP company is saying that the problem is either the switch or the router, though there are never any detectable problems on the network (there is a Windows 2008R2 server …
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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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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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Hi all,
I'm currently facing a strange problem.
I have two Freepbx servers located in different locations.
Location 1 has a sip trunk for incoming and outgoing calls and everything works perfectly.
The second location don't have a sip trunk yet and we're going to implement one soon.
But the local communication is working too.

Both sides are connected with VPN site to site.

I can register a phone in site number 1 into site number 2 PBX and it works.

I can also register a phone in site number 2 into site 1 PBX and it's also works.

My problem is that I was trying to connect both PBX with trunk beetween them, but no matter which trunk I've used (IAX2 or SIP) when I'm trying to make a call from one site to the other site I get "extension is unavailable".
I've checked and the trunk is up.

On both sites I did outbound route (on the first site he is located before the trunk to the sip provider) but still can't get this to work.

Any ideas?
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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.
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Need More Insight Into What’s Killing Your Network
Need More Insight Into What’s Killing Your Network

Flow data analysis from SolarWinds NetFlow Traffic Analyzer (NTA), along with Network Performance Monitor (NPM), can give you deeper visibility into your network’s traffic.

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.
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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.
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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

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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
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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
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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).