IP Telephony




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IP telephony (Internet Protocol telephony) is a general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). Using the Internet, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN. The challenge in IP telephony is to deliver the voice, fax, or video packets in a dependable flow to the user.

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Does anyone know why the switch slows down so much that the VOIP starts to have Jitter and after a reboot it works fine for a week again. The switch runs the PC's through the Phones. There are no VLAN's programmed at the moment. The switch is stock standard. Are there any settings I can change?
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Cisco Unity Express - Version 8.6.7
Cisco Unified Communications Manager Express - 15.6(3)  / CME 11.5

Internal extension to extension calls will transfer to voicemail fine.

Calls from outside that go to voicemail get the message - "There is no mailbox associated with this extension, wait while I transfer your call."  Then it goes to a busy signal.

If the extension is forwarded to another extension internally, calls from both inside and outside will transfer to the other extension.

If the extension is forwarded to an outside number (we dial 9 to get out and enter it into the number to forward to) calls from outside will be transferred to the outside number.  Calls from inside get a busy signal.

Any help would be much appreciated.
Hello all,

I have some Win 2012 3cx v15 phone systems and was having trouble with apple push notifications for calls to remote devices.  I've determined it to be a TLS issue.  I had used IIS Crypto to remove the less secure SSL 3.0, TLS 1.0 and 1.1, leaving just TLS 1.2 and more secure ciphers.  This breaks apple push notifications from the 3cx server/software.  I put back TLS 1.1, no luck.  Put back TLS 1.0, now push notifications work.  I find it odd that I should still need 1.0 enabled on the server.  

Is apple push still using that protocol and not 1.1 or 1.2, or might there be something else going on here.

I'm by no means familiar with protocols/ciphers, just determined what fixes the problem.
sing SIP RTP CoS mark 5
    -- Executing [09599259123@numberplan2:1] Dial("SIP/220-00000015", "DAHDI/G1/9599259123,60") in new stack
[Jun 13 17:19:19] WARNING[3910]: app_dial.c:1745 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [09599259123@numberplan2:2] Hangup("SIP/220-00000015", "") in new stack
Hello Experts. we have Cisco Unity Connection, version: 8.6.2ES44.22900-44, in our Environment. Today the Cisco Unity connection suspended connection because we had exceeded the number of licenses we had available, and as a result, the cisco unity connection suspended connection resulting in users not being able to access voicemail, and auto attendent numbers not working. My questions is, before whenever someone tried to create a VM for a user, and we had no licenses available, it would give error, and not allow for the new VM to be created, but today we saw that it did. Why?, how did it allow someone to create a Voicemail if there were no licenses available, why did it not error out. Please assist in troubleshooting. thanks in advance.
I need for the replacement to look just like these?  Notice that they do not have the Plantronics branding on them.plantronics
I need to create a procedure in my PBX (running Elastix) that hangup some calls depending on the caller ID and Dial a specific phone number to all other numbers.

At this moment I use Goto to send a call to a queue, but I want to use a direct Dial or a MiscDestination.

This is my current code: exten => 4821,n(message),Goto(ext-queues,5555,1)

So instead the "ext-queues,5555,1" I would like to direct dial a phone for example 8775555555

My dial plan requires me to dial 877 before the phone number.

For example in Misc Destinations, I can put 8775555555 and If I use it I will be calling phone 5555555

I have CUCM 10.5 and we use 9+1+(areacode) + number to dial out.  We just introduced Jabber and would like to leverage the dialing from Outlook as well as urls.  

The issue is that it is only the (AreaCode) + Number and it needs the 9+1 in the prefix.
I found something about using \+.! and the PreDot 900  https://supportforums.cisco.com/discussion/11950616/jabber-click-dial-outlook-prefix-9.

I am not familiar with Translation Patterns.  Can someone explain this a bit more.
Hi All Expert,

Good Day.

I would like to check if there is anyway on how to check whether the company is using which phone PBX system?

I have the above phone trying to VPN with a Dell SonicWall TZ400. When I put in the VPN information, listed below, the phone fails and gives me error codes that Phase 2 no response. I will list the three error codes I also see, if anyone can point me in the right direction.


SonicWall VPN Settings:

Policy Type: Tunnel Interface
Authentication Method: IKE using Preshared Secret

IPsec Primary Gateway Name or Address:

IKE Authentication:

Local IKE ID: Domain Name
Peer IKE ID: Domain Name

IKE (Phase 1) Proposal:

Exchange: Aggressive Mod
DH Group: 2
Encryption: 3DES
Authentication: SHA1
Life Time: 28800

IPsec (Phase 2) Proposal:

Protocol: ESp
Encryption: 3DES
Authentication: SHA1
Enable Perfect Forward Secrecy: Checked
DH Group: 2
Life time: 28800

In advanced tab, the only thing checked is Keep Alive.


Server: 50.XX.XX.209
PSK: *****
IKE Parameters: DH2-3DES-SHA1
IPSEC Parameters: DH2-3DES-SHA1
VPN Start Mode: Boot

Password Type: N/A
Encapsulation: RFC
IKE Parameters: DH2-3DES-SHA1
IPSEC Parameters: DH2-3DES-SHA1

Copy TOS: No
File Srvr: Blank
QTest: Disable
Connectivity Check: Never


IKE Phase1 received notify
Error Code: 3997698:18
Module: NOTIFY:305

IKE Phase2 no response
Error code: 397700:0
Module: IKMPD:353

IKE Phase2 no response
Error code: 3997700:0
Module: IKECFG:1184
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Trying to set up internet for home use with the ability to fail over to a second internet connection (such as cellular). Since my phone and security system are both internet based, I just want to see if there is a (cheap) way to ensure device connections stay up if the primary internet goes down. Since the phone and security devices are not tied to a particular external IP, having them suddenly move to a different external IP is not a cause for concern.

I know best practice for business is to fail over to a different ISP, but since this is for home use I'll say that isn't required here.

I'll figure out which ISPs later. Right now just looking for hardware recommends and how it should be configured.
Hi, really struggling with dialplans for Snom 300 IP phone at the moment and would appreciate some help.

I need to set a Snom 300 to only allow outbound calls which begin with a "7" but then to drop the lead digit...  Sounds weird I know, but it's the only way I can think of to restrict outbound calling to the speed-dial list which the users will not be able to view, but able to call.

So the plan is that 01234 567890 (for example) is in the speed dial list as 701234 567890, for the phone to then recognise this as an allowed number but drop the lead 7 so that it is able to be dialled.

I can't change anything in the PBX as we are using a hosted solution - already spoken to them and they can't / won't help, saying it needs to be done at handset level.

I hope that makes sense and that someone can help me out! Thanks in advance. :-)

We have an On-prem shoretel system configured running the director version 18.xx.  We also have three shoretel switches and use both softphones and deskphones.

Shoretel Switches:

Ran through Communicator on PC's

Shoretel IP230

Edge Firewall:
Fortigate 100D

T1 Provider:

New WAN provider:
CenturyLink Fiber

# of Users:

Currently, our phones use a dedicated T1 connection through level3.  This T1 line connects directly to the SG-T1K.  Due to increasingly high costs, we are considering getting rid of the dedicated T1 line with level3 and routing our shoretel phones through our Primary WAN(century link fiber).  The fiber is connected to our Fortigate 100D edge router.  I have worked with shoretel for several years but I have not had to made a change like this before.  

My question is, can we accomplish this with our existing equipment(phones, switches, etc)?  If we can, how do i implement the changes.  If we can not, what needs to change or be upgraded to facilitate this change?

Thank you everyone in advance for any help that you may be able to offer.
I need to move a call manager which is a VM on a UCS C200. Can you please advise
on the proper shutdown procedure and the turn-up? It will retain IP address etc.
Just need to make sure I don't corrupt anything. I see a shutdown procedure
below for CIMC and using the power button. But should I also ssh to Call Manager
first and shut there as well? Thank you.

I have a issue with Dialing out and some inbound.. I have a As5400XM with CT3 card on my TDM side all my b channels are up so are my d channels. My main issue is my dial peer setup when i dial out it will go over 2 dial peers ar the same time and i get no audio. My question is im have 28 T1s and i want them all to do inbound and outbound dialing what is a sample config for some like this. I know a dial peer out going needs a pots dial peer and a voip dial peer just a little confused
Does anyone know if you can dump the call history log of a sip account to a text file?
I need to log voice calls and in particular call-forward calls made by user, by writing in an external syslog, date, time, internal-number, external-number and user originating that.
Which way could I implement this ?

Hi All,
I have been at this all day to no avail.
I am using Yealink IP Phones. The customer now wants to run his laptops with the phones. So the PC's run through the phones.
The phones use their own gateway on port 1 and the PC's use their own on port 24.
In addition to VID 1 created VID 20 for the Data on all ports and Voice on VID 50 Voice as per this example I found.
Phones and PC's are on all the ports except 1 and 24.
AlI really want to do is give priority to the IP Phones.


The phones don't work and neither do the PC's when activated.
I have also setup the phones WAN port with VID 50 and the PC port with VID20.

Any help is welcome
I have not tried tagging P1 and P24 on all 3 the VLANS.  

We are using Cisco Unified Call Manager.

Let's say John wants to call Jane. Both are corporate users but John wants to call Jane from his mobile phone to her mobile phone.

I understand Cisco have a plug in that integrates with Click-to-call API's and allows the creation of an app that performs a call back functionality the voice system will call back John, call Jane, and then bridge the two calls.

Does anyone know the name of the plug in?
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I learning up about Skype Enterprise Voice and it's potential for mobile.

Right now, we use Skype for Skype-to-skype calls. I'm wondering if this can be integrated with the telephone line in the office.  And then, if we can deploy a Skype mobile client to user's personal devices that has their office number....this way, we can :

1. A user never has to give out their personal mobile number, only their office number which will now ring on their personal phone

2. A user can make voice calls, either internally or externally, using this Skype client and reduce cost to them since it presumably uses the corporate network where possible, e.g. for international calls.

Some questions.

1. Is Enterprise Voice the term for the integration of Skype with the phone network

2. When we refer to the Phone Network, do we mean PSTN?

3. Is there a way for an enterprise Skype mobile client to have an external dialler feature so the user can phone anyone, either internal or external?

4. Are there potential cost benefits of this Skype client connecting to the corporate network

5. Is it possible for Skype mobile to have the user's office desk number so that it provides the fixed mobile convergence i talked about? Or does it need a separate number?
We use twilio and sonic firewall in the company for our voip phone system.
And the issue is packet is drop more frequency and do not know what is the issue.

Most of the drops came from inbound calls.

Twilio has no issue.
Plugging a Cisco 7941 phone in to an HP switch. Phone displays "Ethernet disconnected". Phone works fine when plugged into a Cisco switch. I've always had this issue with HP switches and just avoided plugging phones into them but this time its unavoidable. I've checked my Vlan settings and everything else I can think of. Does anyone have experience using HP switches with Cisco phone systems?
  • Have a deployment of 3 servers using dns loadbalancing.
  • we have a trunk setup via an sbc.
  • users have deskphones (CX600)

both inbound and outbound calls work as expected.

however, when a call is placed on hold after 30 seconds the call drops.

the same thing occurs when a call is parked..also after 30 seconds the call drops.

i should mention MOH IS SETUP in this deployment, however when an external call is parked there is just a beep sound.

when a call is parked between users MOH does work and the call does not get dropped.

refer and bypass are set to false on the trunk as well.  Trunk settings

in the snooper log i see references to "this call leg has been replaced"  in the same message as the BYE:
ms-diagnostics-public: 10026;reason="This call leg has been replaced";component="MediationServer"

the trace from the sbc shows that the mediation server is dropping the call so i haven't mentioned that here.

have the snooper trace if needed.

any suggestions appreciated.
How do I go about moving from a phone number hosted with Grasshopper.com to Google Voice?
I have an issue that I need resolved.  We are running Cisco Unity v10.5 and have a problem when people call an extension and it goes unanswered.  If the caller hits "0" it automatically transfers the call to our District Office admin.  I found a setting under the User Templates that had this set up to transfer to the District Office admin and set it to Ignore Key under Caller Input Keys but the problem still persists.  Is there another area that I need to be looking into?

IP Telephony




Articles & Videos



IP telephony (Internet Protocol telephony) is a general term for the technologies that use the Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of information that have traditionally been carried over the dedicated circuit-switched connections of the public switched telephone network (PSTN). Using the Internet, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN. The challenge in IP telephony is to deliver the voice, fax, or video packets in a dependable flow to the user.