IP Telephony

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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We currently have a Cisco CCM system which runs Vlan 10 as the Voice lan and Vlan 2 as the Data, the hosted platform we are going to can have vlan which is set via portal etc, it gets a normal IP and then boots into the correct lan ..... phone will connect ok but will no allow data pass through .... the Cisco phones are working fine but not sure why the hosted wont work ...... any ideas ?

Hi Experts,
I installed CME (ISR 4321 IOS version 15.5), with 40 Phones 7821, 1 IP Phone 8841, another 8851, and 2 IP Conference 8831 using SIP Protocol
all of these SIP Phones work fine, but the call transfer doesn't work, I don't know if some config missing, you find below the sh run,
thank you for your help.
CME-EHM#sh run
Building configuration...

Current configuration : 10874 bytes
! No configuration change since last restart
version 15.5
service timestamps debug datetime msec
service timestamps log datetime msec
no platform punt-keepalive disable-kernel-core
hostname CME-EHM
vrf definition Mgmt-intf
 address-family ipv4
 address-family ipv6
card type e1 0 1
no aaa new-model
subscriber templating
multilink bundle-name authenticated
isdn switch-type primary-net5
crypto pki trustpoint TP-self-signed-246793832
 enrollment selfsigned
 subject-name cn=IOS-Self-Signed-Certificate-246793832
 revocation-check none
 rsakeypair TP-self-signed-246793832
crypto pki certificate chain TP-self-signed-246793832
 certificate self-signed 01
voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
  call start slow
  bind …
In RTMT you can view "Gateway Activity" and choose say MGCP PRI. But the thing that's weird is that it shows PRI channels per CUCM server. Given the description I would think that it would make out how many active calls are happening at the router/gateway not the UCM. Is there a way to get this broken out to you see activity per actual gateway?
Hello Everyone,

My name is George and I was hoping to get some assistance in configuring a Cisco 7914 expansion module.  I have been thrown into the fire with this Cisco phone system.  I have very experience with the CallManager system but I am making some headway.  The biggest problem is that my company is running very outdated versions of the software.  We are running CallManager Administration 4.1 (0.11).  I know, I know...it's no way to run a business but that decision is above my pay grade!  I am told to make it work, as I am sure many of you can understand and appreciate!

So here is what happened.  We had to do a complete network reboot and I was configuring a new phone when they took everything down.  When the network was brought back up, it took a few minutes and all but one phone came back up.  Of course, the one that did not come back up was the receptionist's.  After some troubleshooting I found that somehow, her phone was no longer showing in CallManager.  I went through the process of adding the phone again and defining it and configured.  Her phone works fine now.  The problem is with the 7914 expansion module.

The module is showing all red lights on the buttons and the display is blank.  I went to Add/Update speed dials, entered all the name and extension information and clicked on Update and Close.  I thought this was all I needed to do but I was wrong.  In the Configure Speed Dial Settings for SEPXXXXXXXXX window, it says 'Speed Dial Settings not …
How can you monitor DSP usage on a Cisco ISR used for MTP, transcoding, conferencing? Is there an OID which would let  us graph this?

mtp01-sc#sho sccp conn
sess_id    conn_id      stype mode     codec   sport rport ripaddr conn_id_tx

50630987   251788306    mtp   sendrecv g711u   24918 24576                                                                
50630987   251788304    mtp   sendrecv g711u   31384 18902                                                                  
50631252   50636167     mtp   sendrecv g711u   19210 17128                                                                  
50631252   50636166     mtp   sendrecv g711u   18332 50004    
Total number of active session(s) 11, and connection(s) 22
I am using a grandstream IP PBX UCM6202 and GoIP8 as a VOIP Gateway.
when i dial a number in this pattern +960 7788340 from PBX the call goes out to GoIP gsm gateway but from the GoIP outgoing call has a some unknown character due to space in front of digit 7. for this reason call is not dialed. if there is no space then there is no issue with the call going out. i am using an iphone numbers stored in contacts by defaults has some spaces. is there a way to remove the space when dialled out from GoIP gateway
below is a link to GoIP
While getting ready to move a CUCM cluster I was reminded the route lists associate with a particular CM Group and register to a member of that group. But the question: Why is that necessary?
In Cisco UCM 10, how can I get a listing of all members of a specific device pool?
I have several DT700 to put on my SV8100 switch. I setup all the settings correctly on the phone and can even use the web programming to get to it. What section do I go to for setting it up on the PBX with an extension and such? On the phone's display it shows Full Port.
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?
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.
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
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

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. :-)
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 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.
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?
hi all, my company is going to switch to voice over ip phone system within 1 week. before it is switched, do you know there is any stress testing that we can test our network ensure we can handle all incoming phone calls and etc.?

We have around 50 to 70 voice over ip phones will be used.

We have CallManager (CUCM) 6.1, we want to forward ext. 111 from M-F 4:30 pm to 1:00 am to 555-555-5555.
Also forward ext .111 from M-F 1:00am to 7:45am to (777)-777-7777 and (666) 666-6666.
Can someone help us through it STEP BY STEP.
We are still using Avaya IP Office 403 & 406 control units. Wondering if there is a version of Phone Manager & Softconsole that will work on Windows 10 and with the 403 & 406 control units?
Hi, I'm testing out Office 365 Skype for Business and PSTN calling with Polycom VVX 601 Phones.

I'm familiar with many Hosted PBX Providers such as RingCentral, 8x8, Vonage, etc. where you can just page and intercom to extensions and groups without much configuration on the Hosted PBX Side.

With Office 365 Skype for Business this seems like quite a hurdle. Per this link:


I have to manually go into each phone and enable/disable various channels so that anyone set to receive on a channel can get a page?

Is this the only way to do this?

How about Intercom, how's that setup?
Hi, I have an old door opener system in my building, which lets me speaks when someone rings my department number, and open de door. I'm trying to find a way to control all this with asterisk, perhaps combined with arduino. Not sure if there is such a thing like a door opener with sip conexion, but I'm willing to try to do it myself (with someone else help, obviously :).

My door opener system is like http://www.netyer.com/t3_conexion_basica_con_fuente_generica_4_bornes.html

What I want to accomplish is: When someone is at the door, press my department number and rings in my departmen. Then, I should dial a sip extension, be able to talk, and open de door dialing a code.

Any idea?

IP Telephony

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.