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 doing business networks, I've encountered quite a few VOIP installations.  
Many of them are provided by the ISP provider and run on the same office subnet as the computers.  Even these have troubles from time to time but are really quite simple.
Others have the VOIP set up with a separate internet connection solely for VOIP, provided by a separate VOIP provider, and are isolated from the office network with a VLAN.
In the latter case, responsibilities get blurred.
So, I'm wondering if there is a VOIP industry best practice?

Here's an example:  
One desk's phone often has a screeching noise "like grinding metal" in the audio that is audible on both ends.
The phone has been replaced a couple of times to no avail.
Other phone on the same switch end has no trouble.
I've started a Wireshark capture of the traffic.

Another example:
Calls are sometimes such that audio is only working in one direction.  So, only one participant can hear anything.

We know and understand the network.
The VOIP provider understands VOIP and their system.

What is common and best practice for dealing with things like this?
How much support and service should we expect from the VOIP provider?
Who leads in the investigation?
etc.

I'm not interested in finger-pointing, just trying to calibrate expectations.
I can well imagine that this is contract-dependent but this must be a common situation with a wealth of experiences.
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I am using Freepbx 14 and working fine but I got thousands of attacks and in Intrusion Detection, my public ip  has been blocked sometimes and because of this calls are not working. I am using fortigate firewall and opened the 5060 to 20000 ports for the FreePBX so My question is 1. are ports forward mandatory for inbound route ( if I change the sip registration port from 5060 to other and do same with the trunk provider ) . Please let me know how I can make this FreePBX more secure so call disturbance would not occurred in future.
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Good afternoon everyone.   I was wondering if anyone had any experience using the My Fax application.   Currently one of our customer is using the service to send outbound faxes directly from their EMR application.   The majority of the fax numbers associated with the EMR application they send to are coming through without a problem.   There is one fax number (which I believe to be an analog line) when sending , the EMR application shows the fax sends and completes the send however only partial pages are coming through.   This is only happening with 1 fax line out of the rest.   According to the vendor they did state this.

"Good afternoon. Pls note that this change reflects ALL faxes sent. In our experience, this raises a few additional issues:
 
1) Additional Support Calls in to our team as to why the failed confirmations are delayed (ie could take up to 30 minutes to produce a failed confirmation)
2) If the line we are dialing is of poor quality and cuts off multiple times mid transmission, we will still try multiple times to deliver that fax meaning that they would receive partial faxes…."

Where specifically can I look to find out where the break is happening?   Is there a way to verify if the line quality is poor?  Or if not, and the fax is making it to the destination could the drop happen on the way they are processing the inbound transmission on their end?

Here is some additional information from the vendor..

I have investigated and tested the number …
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Hi,

We have a Skype account for our company.

We sign in with our company email address to access our Skype account on our Windows 10 Pro PC.

For people to get in contact with us, they usually request a Skype ID.

Our Skype ID displays as something like the following: live:" "_(numbers)

Other companies have their Skype Name/ ID displayed without the "live:" part like "walmart" for example.

We would like to change our Skype ID/Name to something better as it does not look professional.

We are aware we can change the display name for the account.

How do we change the Skype Name/ID for our Skype account?

Thanks,
Robbie
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I have Asterisk version 1.8.32.3. Sometimes my recordings have the audio of the agent and the audio of the client desyncronized. I want to know how could this be resolved.

I tried put jitterbuffer in sip.conf, and I changed to res_timing_dahdi.so

Best Regards.
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Issue: Some SPA502G Cisco phones freeze without any warning,

Some users have found that their phone does not work and must restart it to recover it.
About 20 cases reported in the last two weeks (before this had not happened). We have almost 300 devices spa502g.
The trigger of this issue was not found, so the scenario cannot be reproduced.


SPA502G
Software version 7.5.6a
Hardware version 1.0.4

Platform
Freepbx 2.11.0.43
Asterisk 1.8.7.0

No recent updates have been made.
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Hi All,

We have an older Mitel 3300 using predominantly Mitel 5212 phones. We have a dedicated VLAN for voice traffic, which is tagged on our switches alongside whichever VLAN is providing the network for the computers. We are having a strange problem in which when a phone call is made, one side is unable to hear the other (in reverse it is fine). After a few seconds this will sometimes resolve itself, or they will need to hang up and try again. On the retry both sides can usually have full conversation as normal. This happens randomly throughout the day to different number / handsets that are plugged into various switches throughout the organisation. We have had a look at the switches and they don't have any obvious errors / are not reporting they have maxed out on throughput.

We have had our support engineers look at the phone system and they cannot spot anything obvious with regards to the fault.

Any clues as to where to start diagnosing this.

Cheers,
Paul
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Hi,

Can anyone help me figure out the easiest way to configure an EdgeSwitch 24 250w for VoIP QoS?  There is surprising little clear/concise info out there on how to do so and the support I'm seeing for Ubiquiti products has me wishing I would have gone with a different brand.  /miniRant

I have an IPSec tunnel connecting two buildings, the 'remote' building has QoS configured on the Fortigate router, but the switch is basically in default mode.  I have 7 IP phones on site and we are having intermittent quality issues, so QoS on the switch is step one in my problem solving.  Browsing around the gui it looked like the OUI based method would be something I could fight through, but it's not quite that simple after all.

I'm not sure I understand what the OUI is...I though just the first half of the MAC.  The phones are Avaya model J169, and Google tells me the OUI is 00:04:0D, but according the client info from the switch, all of the phones MAC addy's start with C8:1F:EA, so wouldn't that be the OUI value?

Do I still need to create a VLAN with this method or does the 'auto-voip' setup take care of that for me?

Obviously a little over my head here with new stuff, but still disappointed that there isn't a 'how to' I could find...this has to be a very common request, no?
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Recently I have migrated the 3cx on-premises to Cloud and all Ext. are connected through SBC. All inbound and outbound calls are working fine except the voicemail....I am not getting any voicemail after migration and in log I got Main line SBC:Unavailable  .......Internal voicemail are working fine.
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Dear Experts,

We have two Fanvil C600 phones in the office.

One is working well, the other is just showing a blank screen with the led indicator "red".

I have done firmware upgrade (pressing back and home screen button) and reboot the phone but still showing black screen.

I am not sure of wiping out the config as i have not done any backup.

Connected phone to network, the wan ports says "1000 mbps" but I am not able to ping the static ip address

The phone displayed an IP 192.168.1.179, but I cannot ping it.

Connected to lan port of phone but there is no connection.
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The user lives in an apartment complex that provides phone service as part of the rent. They were able to connect their Panasonic wireless home phone with 3 stations to this system.

However, even though the wireless phone receives calls, the caller ID isn't working.  Most inbound calls show the ID as DID/DOD.  Outbound calls show the town or just the phone number.  I called my iPhone, and my iPhone recognized the number as being on my contact list, and thus showed the name.

In addition, the answering machine built into the main station usually doesn't pick up.  Management said that an answering machine should work.  The Panasonic HAS an answering machine.

Is there anything we can do to get the caller ID and answering machine portion of the Panasonic phone working?

Thanks.
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Hi All,

We are looking for an analytics product to report on our usage of our on-premises Mitel VoIP system and Skype for Business Online. Can anyone recommend a product?
Reporting for Skype for Business Online is the priority, but we will likely need both in the future.
We are a medium sized charity and so our budget is limited.

Cheers, Ian
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I am using 3cx to connect with twilio.

I know that 3cx normally uses Elastic SIP Trunking, but I need to use programmable voice.

I have everything working as I think it should except when making an outbound call.

If I use an app directly connected to twilio then it's fine, but when routing through 3cx via the app or desk phone I get an error.

Error - 32009 The user you tried to dial is not registered with the corresponding SIP Domain

The logs show that 3cx is trying to place the dialed number in sip:+1##########@name.sip.us1.twilio.com

When it should be sip:username@name.sip.us1.twilio.com in order for twilio to be happy.

I've tried changing settings in 3cx to use the AuthID when sending out and I think I even got it working for a moment, but when I tried to repeat the process I couldn't figure it out again.

So does anyone know how to use programmable voice with 3cx?
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Say, the following device :  D-Link 3G FLLA Wi-Fi Model DWR-720/PW   takes a SIM card and allows one to mkake telephone calls from the built-in handset. It also offers wifi to the user. We are loking for a device with similar functionality as described, except that it does not need a handset but rather a FXO Port allowing one to connect in a regular POTS.
The important features are the Wifi for the end user and the ability to plug in a POTS. Cost is a factor and availability in South Africa is preferable. Kindly suggest devices ofering this functionality.
PS.  This device is being used as a replacemenent to a traditional analog telephone
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I'm working on an Allworx Server, I'm trying to get a Handset Template to become active.

Can anyone tell me how to get that marked as active?

HandsetTemplatesPage-1-.png
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I am trying to setup a new Asterisk v13 machine to accept SIP connections from an oldr set of machines which are still running Asterisk v11.

When one of the older Asterisk 11 servers sends a call to V13 I get the following errors :-
[Jul 23 16:52:50] NOTICE[23105]: res_pjsip/pjsip_distributor.c:666 log_failed_request: Request 'INVITE' from '<sip:gw@1.1.1.21>' failed for '1.1.1.21:5060' (callid: 0683aab831dd1bd31fb332de52eaf134@1.1.1.21:5060) - Failed to authenticate
[Jul 23 16:52:50] NOTICE[23105]: res_pjsip/pjsip_distributor.c:666 log_failed_request: Request 'INVITE' from '<sip:gw@1.1.1.21>' failed for '1.1.1.21:5060' (callid: 0683aab831dd1bd31fb332de52eaf134@1.1.1.21:5060) - Failed to authenticate
[Jul 23 16:52:50] NOTICE[23105]: res_pjsip/pjsip_distributor.c:666 log_failed_request: Request 'INVITE' from '<sip:gw@1.1.1.21>' failed for '1.1.1.21:5060' (callid: 0683aab831dd1bd31fb332de52eaf134@1.1.1.21:5060) - Failed to authenticate

I have attached the relevant parts of the pjsip configuration. As I understand it the type=identity line should match the IP address and point the incoming SIP to the 'gateways' endpoint which is then configured to use the intergateway_auth authentication.
pjsip-config.txt
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Voicemails showing up are dated 2 days in the future...

I just took over IT for a school that is using Avaya IP Office (Avaya VoiceMail Pro).  Both the dates on the Avaya server and the Voicemail Pro server seem to be accurate (today).  But when a v'mail comes in, on the phone, it shows the data as 7/17 (two days from now).  See attachment:

Can't seem to figure it out.  Any help is appreciated...
Screen-Shot-2019-07-15-at-1.22.54-PM.png
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We have Avaya IPOffice 500 V2 system in HQ.

We are building remote location in other state and will have Point-to-point connection.

That location only have 10 users. I would like to give them IP phone via P-to-p connection.
What equipment I will need and how can I make them connected?
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My current setup is this- I use a Watchguard firewall.
Interface 0 is external.
Interface 1 is trusted-192.168.1.1/24
Interface 2 is trusted-192.168.3.1/24
There is a VPN to another office that is 192.168.2.1/24

Our phone system is 192.168.1.5
If I plug a phone into the .2 network the phone will connect up without an issue.
If I plug a phone into the .3 network the phone will NOT connect up.

I assume there needs to be a policy in place to get the two to talk. I am unsure of what the policy needs to be.
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Hi,

We have around 7 Yealink T22p. 

Phone quality is good when only one phone is being used.

If there are more than two phones are being used at the same time, one of them breaks up a lot but it breaks up the quality from our end. It means that I can hear the other person well but this person cannot hear me well as my voice breaks up.

They are all connected to a Cisco POE switch (2960-S). We have NBN Internet 80Mb/30Mb.

Summary, 
1. One call at a time is okay 
2. If two devices are making calls or being used at the same time, one of them breaks out a lot. The test was carried out when the network is free during the weekend.

The test was carried out when the network is free during the weekend.
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I would Like to change my cloud and sip voip  provider  from ubity Cloud  IP Telephony Provider to Avaya IP Cloud Provider
As I have Mikrotek Cr 1016 as vpn gateway and HP voip switches and polycom vvx 300
what are network requirements  and ip telephone models for having avaya cloud considerations for avaya experts
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hi - its a general question trying to understand how phones/phonesystems works .

got a client who uses nec phone systems SV9100 and nec phones.

connections are made like,:
phones are connected with telephone cables and going to patch panels
from patchpanels - using ethernet cable connecting to another patchpanels which has extension numbers.
extension numbers are connected back to phone systemsSV9100 in digital station interface and single line interface . (black cables in pic)
from phone system SV9100- a voip port is connected to lan switch (grey cable in pic)
and got another device "one access"  it has a ethernet cable connecting to lan switch.

i knew am confusing. but just need general idea- what is digital station interface and single line interface in sV9100 and what one device device does in the network ???
IMG_5616.jpgIMG_5617.jpg
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I have a brand new out of the box Grandstream UCM6510 that is refusing connections to the GUI so we are unable to set it up. We can see the IP address it has picked from DHCP on the LCD however it is refusing connections on that address. We are connected on the WAN port as per the instructions.
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Hi I'm trying to setup a user on Avaya IP Office 6.2 to send voice mail to email.  The emails work for other users, but this is a user that is taking over an existing extension, so I need to modify (I think) the existing extensions voice mail to email.

Thanks all
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We have a Cisco UC520 - all is working fine, landlines connected, extensions connected etc. etc.

We want to add our VOIP account to the system. We have the details given to us by our provider, these are as follows:

1 Make sure that your SIP Phone is turned on and connected to an IP Router or Modem. If you are using a softphone make sure that your PC is connected to the network.
In your phone's configuration menu there should be an option to define a SIP Server, SIP Registrar or SIP Domain value. Set the value to: voiptalk.org.

2. For SIP Authentication, set your SIP User ID or similar to your VoIPtalk ID (eg 84411076) and set SIP Password or similar to your six-digit VoIPtalk Password provided in your activation email.

3. If you have an Outbound Proxy setting, set this to: nat.voiptalk.org:5065. Alternatively, if this setting allows you to define the port in a separate field, set the IP Address to: nat.voiptalk.org and the port number to: 5065.

4. Confirm the settings by reloading or rebooting your phone. Dial 902 to confirm your configuration.

5. If you encounter any problems, make sure that your IP Router or ISP is not blocking any traffic, specifically on ports 5060 and 5065. Also ensure that the router is not blocking UDP traffic. You can contact your network administrator or ISP for more information.

6. For additional assistance please refer to our instructions for other VoIP phones. You can also contact our support team via email. Please state your…
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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).