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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having issue with our VOIP.  We have a connection via SIP from our remote location back to HQ.  We lost connection all of a sudden and verified with our provider that everything is up on their end.  We replaced both the the t1k and 50v device and also replaced a Meraki device with a new cisco switch to bypass the meraki.  I don't know where to go from here.  There was a vlan change the week before where, I was told, two VLANS were consolidated into one.  Can you point me in the correct direction.  I'm in the Palo Alto Firewall now and thinking that this is where the issue might be...I think I pretty much checked everything, including QOS.  I do see in the logs that there are messages about insufficient data under the application and also see aged-out under the session end reason.  This is under Monitor/Logs/traffic.

We have a shortel VOIP system.
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Hi Experts,

We are moving to a new Mitel Cloud service and have IP485G phones.

I setup a VLAN on our switches along with a DHCP scope on a windows 2012r2 DC.

We receive the correct IP, router, dns  from the VOIP VLAN but the option .156 is not populating in order to get config files and firmware updates.

I am using the string below configServers="update.sky.shoretel.com"  

Thank You
 I am looking for an internet router with dual WAN ports.  I use CISCO C891F-K9 and it works well. But it only delivers up to 100Mbps speed in one of the WAN ports. Recently Spectrum installed 400/20 Mbps internet speed and I like to make sure that both WAN ports can handle internet speed up to 1000Mbps.
  I love CISCO "high-end" product simply because it has not failed or died during my ownership. I have retired them because technology was old, but never because of mechanical failure. However, I am not limited to CISCO brand, so I like to hear about other products.
- has to be able to handle VoIP phones
- Dual WAN ports with 1000Mbps speed
- Router must be able to switch between primary and backup WAN ports on-demand automatically.
- 24/x7 tech support with annual support fee paid.
- Once configured and installed, you can forget about it.
- does not have to have GUI

Thanks in advance.
Several clients have expressed an interest in getting a blocking service to prevent junk phone calls from being received on their land lines.

Is there a filtering service that can work with VOIP provided by  Xfinity, FIOS, Optimum Online, and other providers?
Is there one for those using copper wiring?

I have been using RoboKiller on my iPhone, and I've been happy with that service.
Something like that for land lines would be great.

I have a client 12 staff that had a 800Kb/s ADSL service that was carrying LAN traffic, EMail, web surfing, and VOIP.  Call quality was bad but thats how the Phone provider deployed it.

On the weekend I replaced their ADSL with VDSL and we not get approx 48Mb/s.

As expected, outgoing/incomming calls are massively improved, as is web surfing etc, but internal calls have deteriorated further.  Calls struggle to be audible.  I am Stumped as to why this is happening
While trying to apply a QoS outbound rule to a Catalyst 9300:

Invalid queuing class-map!!! Queuing actions supported only with dscp/cos/qos-group/precedence/exp based classification!!!

The config we are using is below:

policy-map Shoretel-Input-Policy
class class-shoretel-media-input
  set dscp ef
class class-shoretel-signaling-input
  set dscp cs3
class class-default
  set dscp default
policy-map Shoretel-Output-Policy
class class-shoretel-media-output
  set dscp ef
  priority level 1 percent 5
class class-shoretel-signaling-output
  set dscp cs3
  bandwidth remaining percent 15 
 class class-default
  set dscp default
  bandwidth remaining percent 60

Open in new window

Running into a problem with CUCM when doing an export of Phones with Specific Details.  I have several phones that have 15 lines (18 lines in the button template total) on them but when exporting, the 2nd-15th lines don't show up until Line Number 30 or later.

What would cause CUCM v11.5 to export the phones lines like that?
Product: Outlook 365
Scenario: Calls that comes to the mainline (main phone number) should go to a separate inbox, let's call that the main number. However, calls that come to both the main number and a personal mail box or rings on the same phone of another user. In other words, two numbers are assigned to the same IP phone.

Goal: I am trying to prevent voice mails from going to the personal inbox, which are somethings happen. The voice mail should only go to the main line inbox.
Other facts: When a call is sent to the main number, Skype for Business receives the call on the personal Skype for Business user's account.
The delegate phone number is setup in Outlook as an additional mailbox.
Calls to the mainline are never picked up, but is utilized only as a voice mail box service.
How can I prevent voice mail from going to the personal inbox, in Outlook 365?
I have a Cisco 7960 IP phone and wanted to verify if there is a way to connect the phone via my home network without having any Cisco equipment?  If so, how can I use the phone to make calls or receive calls?
I wish to setup On-Premise IP-PBX Business on my local server using proxmox vm. While being able to sync with an on premise Crm like bitrix24

I was thinking of getting switchvox. Kindly advise.
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I have Cisco Call Manager 11x with Unity 10x

Once in awhile I see an issue where I dial a users extension, the phone rings, they don't pick up, and instead of kicking me to their Unity Greeting to leave a voicemail it kicks me to "Your call cannot be completed as dialed. Please consult your directory and call again or ask your operator for help."

I wait 1 minute, dial, it works. it's like call manager doesn't forward it to unity.

Is there a limit on how many connections Unity can accept from Call Manager simultaneously?
We have an Allworx phone system, with HP and Aruba Procurve (PoE) switches connecting the phones and servers. We have a problem with intermittent one-way audio- I think I know the cause but I don't know how to solve it. Our phones are supposed to obtain IP addresses in the range from the Allworx server. I have found that a small number of phones randomly obtain IP addresses from our Windows DHCP server in the range The VoIP VLAN is set to 200 and the Windows distribution VLAN is set to the default (I believe it is numbered as 0 on the switches). Every time a phone has the one-way audio problem, it is in the Windows distribution range. Not every phone in the windows range has the problem, but every time a phone has a problem it is in that range. We don't want to expand the VoIP traffic into the Windows range. But I think the problem is the phones that get the wrong IP addresses. It is not limited to a set of phones, but there are usually about 5 or 6 that could be any of 45 phones at this site. All phones are statically set when they are issued to use VLAN 200, and the problem occurs regardless of phone or switch port. Users are not moving equipment around, so that is not the cause.

So the problem is that DHCP appears to be bleeding into VLAN 200. The PC's on the DEFAULT VLAN never get IP addresses in the network. Switches are up-to-date with firmware. We have this problem at other sites as well, but one site …
We recently started noticing that a certain floor of a building on campus can only call 1 direction.  When either side tries to call the one side in the bottom of our building cannot hear anything that the other side is saying. This only happens to the 3905's on the first floor of a specific building. If you use a higher model phone on that floor it doesnt happen and if you bring a 3905 up a floor or anywhere else and it functions perfectly.  Strangely, this isn't consistent as some 3905 phones on that floor can call some phones in other areas of the building and there are no problems at all.  I am stumped.  Any suggestions would be appreciated!

Phone system is Cisco
Switches are Cisco 2960S-48fps-L models with IOS 15
Call Manager and Unity versions are 11.5.1
Phone affected on the first floor of the one building are Cisco 3905's

If any log files are needed I can provide those.
Please recommend an on-site call recording solution for a UK based call centre. We are using Avaya IP Office 11 phone system (IPO control unit 500 v2).

It needs to be sold and properly supported in the UK (even post Brexit).

We are having major issues with our current solution and are seriously considering a migration to a new product. I'd prefer any recommendations based on honest, hand-on experience rather than a sales pitch.
We have a voice gateway at a remote location currently directing outbound calls over a pair of POTS lines. The location is very small but even so E911 compliance requires each of the 4 handsets to identify their unique location for caller ID.
I am looking to redirect outbound calls from this location back over the ELAN and out through our Call Manager/SIP connection as we do with other locations. All phones/locations/SRST etc have already been set up in CM, so I suspect this is just a matter of directing the calls.

Can someone give a brief/broad pointer of where to look (in the VG or CM)? At this point if I have the basic starting point I may be able to compare to other locations to figure out, nothing is jumping out at me presently and I'm unsure which settings influence or are irrelevant.

Thanks in advance!
We are moving into a new building, and we will have all newer Cisco switches, 2960X and 3850's for the cores.
I'm planning to have different vlans for the servers, PCs, VoIP phones, but I was thinking, since all of the different equipment need to communicate with the servers,
I will need to allow and route all the different vlans to access the servers vlan.  If that's the case, is then better to just create one flat network, everyone in one vlan, a /22 instead?
I guess I need to find some good articles on line to dig deeper into vlans, but on the surface, besides having a smaller broadcast domain, it just adds more complexity.

Any thoughts?

My company currently it's moving to a new phone system and we are stock. our DHCP it's set to IP Scope 192.168.16.xx and I created a second Scope 10.11.0.xx so it can connect via VPN tunnel with the VoIP system of our another office (we are in So. Cal and the other office in Florida) now, To my knowledge I need to create the scopes and the services on DHCP so I can setup the relay to ensure that traffic can go from the 10.11 network using the 192.168 network as gateway and at some point  create a VLAN in my switches to route.

I did all the first part until before the VLAN part, I have some problems.

1-Computers on my Scope 192.168.16.xx are registering on the 10.11.0.xx I need to know how to stop them from doing that, I need to keep them alive but without merging

2-Do I need to create a vlan to route all my VoIP traffic ? we have layer 2 switches and the router it's managed by our ISP or Do I need to setup a a new port in my firewall with that subnet routing all traffic from 10.11 to the public IP

I have a VM running server 2008 R2 as my DHCP I have 2 virtual NICS installed one running on 192.168.16.xx and the other on 10.11.0.xx
I have RRA installed with IGMP installed, and my gut tells me that I did something wrong

I have not done something like this in years so if there is anyone that can give me some guiadence I will really appreciate it.
Being a network administrator, among other things, I'm often asked by users to open ports in a firewall.
Usually the users don't know much about what they're asking for so they can't answer any questions - just forward what their technical people have provided.

Here is a typical example for a VOIP system:

The full network information for the VoIP system is:
Port Range (Audio): 35000-65000 UDP
Port Range (SIP): 5060 UDP, 5061 TLS
Port Range (Configuration Servers): 1024-65536 TCP source port, TCP Destination ports: 80, 443, 1443, 2443, 6716,
Port Range (Presence Servers): TCP Destination ports: 5222 and 5280.
I guess that's all well and good if you understand the context but that's where I'm not the expert.

I can set up firewall rules but, being conservative, I don't want to open incoming ports just willy-nilly in order to assure that the requestor gets what he/she wants.
If I ask them: "Are these incoming ports or outgoing ports?" they have no idea.
In some cases, I'm sure that some are outgoing.....
What I'm used to, for the most part, is that all outgoing will be allowed and all incoming will be blocked unless initiated by outgoing traffic.
Given this limited view, I would want to set up to allow incoming traffic to certain ports and leave things at that.
But, which ones?

I know this is likely a naive question.
So, in my context of understanding, how would you interpret the specification above?
And, in the details, I've never set …
I had this question after viewing "Incorrect Username or Password" on log in.

After setting up a new VoIP phone system from Comcast Business on our network, which required re-configuring our Dell network switch with VLANs for voice and data, we started to see issues with users not able to login to the network even though their credentials are valid. I would like to know if others have a similar experience and if so what is the best solution to avoid this kind of problems. Also, I am still trying to resolve the login problems for the users and the only way I have been able to use thus far is to have the user reboot their PC and then they are able to login again. I had similar problems with my domain admin account randomly on different servers. Why is it that on some servers my login works and others it does not?
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Hi All, I am attempting to configure an AAPT IP based trunk in 3CX via a dedicated TPG SIP service, and am struggling to get it working.

What I really am after is examples of working AAPT trunk configurations that I can compare my set up to.

If anyone out there could provide some examples of correct trunk configuration, I would be extremely grateful.

Wireshark shows OPTIONS messages successfully hitting the phone system from TPG SIP server, and 200 OKAY messages being sent to TPG SIP server.


 - 3CX on premise
 - Wireshark shows OPTIONS being received and 200 OKAY being sent to/from TPG SIP IP
 - Dual WAN connection with dedicated SIP service on WAN2
 - NAT on WAN2 to local IP of 3CX
 - Static routes are configured to route required SIP traffic in/out via either WAN1 or WAN2 depending on what port is required

Kind regards,


I have a problem with an asterisk server:

I have a SIP trunk from Vodafone. When I call from another provider ( lets say Orange ) redirecting from Softphone to another extension works. When I call from the same provider ( Vodafone - my sip trunks use vodafone ) and try to redirect from the softphone to another extension, the call is intrerupting.

This is extension.conf related to extension number used for redirecting: (I masked the real number)

exten => _yyy268,1,Set(CALLFILENAME=${CALLERID(num)}_${EXTEN}-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)})
exten => _yyy268,n,MixMonitor(/var/inregistrari/in/${CALLFILENAME}.wav,b)
;exten => _yyy268,n,Goto(ivr-liber,9,1)
exten => _yyy268,n,GotoIfTime(18:00-23:59|mon-fri|*|*?time,5692,1)
exten => _yyy268,n,GotoIfTime(00:00-08:00|mon-fri|*|*?time,5692,1)
exten => _yyy268,n,Dial(SIP/268,20,tk)
exten => _yyy268,n,Dial(SIP/241,60,tk)
exten => _yyy268,n,Congestion()
exten => _yyy268,n,Hangup()

THis is sip.conf related to extension used for redirecting:



THis is sip.cinf related to the trunk used:


We are having an issue where we receive a call that rings all phones in our ring group with a 3CX phone system. When a user answers the call, no one is there. The call disconnects, then the next extension in the ring group will ring. The phones will continue to ring until every extension in the ring group answers the call.

There is no caller id showing on the phones, and no one on the other end of the call. Furthermore, the issue is reoccurring once every week to two weeks. It happens on a random day and time. We are not able to replicate the issue and is affecting clients that do not have a IVR.

We've open a ticket with our SIP provider and 3CX. Our SIP provider says that when this occurs, a SIP cancel is sent to the 3CX PBX but the PBX does not respond and the call gets caught up in the system. 3CX says it's a know issue with the phone system but has been over 2 months with no resolution. Any help would be appreciated.
I have a big problem with Cisco voip configuration. I have two CME router which is connected by IPsec over gre tunnel vpn. The flow between router Irak and router CI is correct but we can not make any calls, we heard a busy tone. However the calls between Router CI and Router Irak work well.
I don't know how to fix this issue. I need your help please.

Best Regards,
I'm wanting to setup QoS for Skype for Business Online. When I do a packet capture I see that the real time
ports that come into play are the UDP 50,000 - 59,999. The article below calls the 50,000 - 59,999 as optional.
Is there any way through group policy to tell skype to use the UDP 3478, 3479, 3480, 3481 only or at least
to prefer it? Marking all TCP/UDP 50,000-59,999 for EF classification seems pretty broad.

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!

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).