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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VOIP calls suffering from silent patches.

We had BT Cloud Voice put in around 6 months back and have suffered with issues since it was put in. We put this down to our FTTC lines, however we have now had a Virgin Leased Line installed (100mb/1gb).

We currently have around 90 computers and 90 phones. These are run through Netgear GS752TPv2 switches and to a Draytek 2926 router.

Looking on the router, the syslog is showing quite a lot of errors similar to the one below:

 [VoIP_QoS] VoIP RTP[45348] Rx Loss detected: 16/115 ...

There is also a few errors as below:
 [VoIP_QoS] VoIP RTP[45344] latency detected...220.

I'm assuming the first error is the receive packet loss (of around 15% in the case above) which could be causing the silent patches.

As well as this, I've setup a ping to the router from two different computers that are in different switches. They are both getting timeouts at the same time, one or twice a minute (sometimes there is a longer gap, sometimes there will be 2 within 10 seconds).

Any ideas where to start looking to try and resolve this issue?

Thanks, Shane
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Hi,

We have some  voip software called unified communications UC. It suddenly stopped working for authentication back the the Service provider (but only doesnt work when on our LAN going via the ASA 9.0(2) . If hot spot by phone or any other method its fine.

Ive checked all ports to be open - in fact tried opening all outbound ports from my computer - same result. - If I do a wireshark ip.dt == to the Service provider SIP endpoint then try log in to the UC software I see totally nil  SIP traffic to that IP in fact nil sip traffic total. when i  run same wireshark when hotspotted to phone on same device I see the registration sip successful. ? all is fine

ive tried turning off and on sip inspection on ASA many times - doesnt fix. strangely however I did ONCE get a successful registration - and then never again!!! same failure to do anything even nothing in wireshark for sip. I cant see any denies on asa monitoring as well. ACL's all in place. Im a bit los from here service provider saying asa fault. please help
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I have a VoIP phone with an OpenVPN client.   There is an issue when used on OpenVPN, so I want to inspect the data, SIP trace etc to help find the problem (fast then then the vendor)

My problem is the data I want to see is wrapped/encrypted with OpenVPN.   I have the private key and client certificates, therefore I thinking it must be possible to decrypt the capture and find the information I'm looking for?

How can I use Wireshark (or other opensource/free tool) to achieve this?

I have reasonable experience with Wireshark, used not used it to decrypt traffic before.


Many thanks


OpenVPN pcap
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Looking for ways to limit users to only using Microsoft Teams for IM, voice and video calls.

We already use Slack and think that Slack does what Slack does better than Teams.

Unfortunately due to Skype for Business going end of life, we are being forced into Teams.

Some of the telecoms, video-conferencing and IM stuff is great in Teams but we prefer Slack for collaborative working.

We don't want to confuse users though, so are looking for ways to lock down the Teams specific stuff but still allow people to use it for the stuff it does well.

Has anyone got any advice about how to lock it down?

One other way we thought we could tackle it was to have notifications enabled for anyone that creates so that we can delete them / advise against.  Any thoughts about achieving that?


Jon
0
Experts,

I'm trying to make calls from my Outlook client. I in order to achieve this I needed to signup with a provider that provides SIP Trunking.

I've signed with a company called Dial 9 https://www.dial9.co.uk/

I'm now trying to set up my Outlook using the following link

http://www.voip-connections.com/contacts/how-to-dial-contacts-from-outlook/

Everything, so far, appears to be going in the right direction, however when I attempt to configure the TAPI settings it is greyed out .. see image

modemThis would suggest that I need to access the Phone and Modem setting with Administrative rights, but I'm not given the option to access using admin rights.

Can someone let me know where I'm going wrong?

Thanks
0
Afternoon,

So i've got an odd one, we're using Skype for business 2016 along with Skype for business online, a lot of the time there are no issues at all however we're getting calls being passed from reception to users whom have call fowarding on and it drops the call.

My knowledge of this is limited and as such would love some direction on what to look at, where to go and what I can try to do to sort this out.

Thanks
Alex
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Hi,

I have a Samsung Note 9 phone.

Sometimes I would get a call from someone while being in a call already with another person.

The person trying to call me never knows that I was in a call with someone else so don't ring back.

To the person trying to ring me, their phone rings for a bit and then goes to voicemail.

I would like to have the person trying to call me in this situation given a message saying "person in a call" or busy.

Is there a setting for this on Note 9 phone?

Thanks,
Robbie
0
I am looking for a way for users/extensions to have the ability  to login/logout their particular hunt groups (on-demand) in CME 8.6. The main purpose is to have all calls forwarded to an answering service when all users/extensions are logged out the hunt group

Has anyone setup something like this?

Thanks!
0
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.
0
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?
0
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.
0
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
0
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?
0
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
0
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
0
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?
0
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.
0
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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.
0
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
0
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
0
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.
0
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
0
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…
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).