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.

Skype for business 2019 Yealink T58A | Trusted Certificate
I need assistance with phones we recently purchased, all T58A handsets with Skype for business firmware.
I have tried all 3 latest available firmware and stuck with this version as it offered a simpler login screen for users.
Phones register correctly for as long as the trusted certificate is not present.
Periodically the handsets will populate with a CA certificate on line 1 even though everything is set to disabled below and then the users are unable to sign into the phones.


What is causing the phones to download the internal root domain CA certificate?
I have inherited a couple company IP sisco phones. I am thinking about selling them as they still have a good used resale value.
Does anyone know if the serial numbers will be registered somewhere, that will cause me grief if I resell them?
Is a serial number necessary to setup the phone on a network? Should I remove the serial or is it needed?

Thanks so much.
I have setup Enterprise Voice / Direct Routing for Microsoft's 'Phone System' element of Office 365 and Teams. We're using a Ribbon SBC Edge (1000) border controller which has a very useful wizard and it's own FQDN certificate from godaddy.

I've enabled enterprise voice in Office 365 online powershell and setup routing policies to allow outbound calling to +44 uk numbers.

In Direct Routing, the SBC is showing as active and in use. I can make calls TO MS Teams clients using DDIs on our SBC routing calls successfully to Teams. However, Teams clients cannot dial out to the PSTN.

We've run trace logging on the SBC which shows all the SIP traces - it can see all the 200 OPTIONS coming to and replying from Microsoft, but it never sees microsoft attempt an invite to a call. For some reason Teams is not reaching out to our SBC when we try to make a call.

Does anyone have any advice for troubleshooting in O365/Powershell to determine why the call doesn't reach our SBC?

We have CUCM and all calls are showing up as the main number. We have configured that mask in the DN. When I do the call analyzer this is what I get,

Calling Party Transformations
External Phone Number Mask = YES
Calling Party Mask =
Prefix =
CallingLineId Presentation = Default
CallingName Presentation = Default
Calling Party Number = 7033

I checked the route pattern configuration and the mask box is checked, The route list detail use external mask is set to default.

 I am missing something but I do not know where. Any help would be appreciated.

Results Summary
Calling Party Information
Calling Party = 7033
Partition = ENT-EXT:Admin-SVC:Admin-xxx-Paging:Admin-PSTN:ENT-SVC:ENT-VM
Device CSS =
Line CSS = Admin-DEVICE
AAR Group Name =
Dialed Digits = 81xxx2
Match Result = RouteThisPattern
Matched Pattern Information
Called Party Number = 81xxx2
Time Zone = America/Chicago
End Device = xxx-TF-LD-RL
Call Classification = OffNet
InterDigit Timeout = NO
Device Override = Disabled
Outside Dial Tone = NO
Call Flow
Route Pattern :Pattern= 8.@
Positional Match List = 8:1:xxx:xx:xx2
DialPlan = North American Numbering Plan
Route Filter
Filter Name = LongDistance
Require Forced Authorization Code = No
Authorization Level = 0
Require Client Matter Code = No
Call Classification = OffNet
PreTransform Calling Party …
I am building the system and I am not sure sure in my architecture of the future system: The system enables people call to companies and organizations though web browser using WEB-RTC. Use-cases are following:

FROM browser to HOME PHONE
FROM browser to mobile phone number
FROM browser to browser
FROM browser to existing corporate call-center (IP)


Since all calls are recorded, there are different options how calls might be implemented: peer to peer, through media server. Also, there might be more than 1 million concurrent calls. System parts should be free/open source. In my understanding, system design is following(attached). However, i am not sure in this. Please, help to build right architecture.

I'm trying to connect (send command)  to an IP phone (Yealink) from outside the network.
It's already working from the inside with this simple URL to call:
It's very easy and can make the phone call a client directly from my custom app (Access database).

I see that IP phone company have remote access to my IP phone and would like to know the Technic to do the same.

I have users using the custom app on a remote server, but the IP phone is at another location, and I want to make there phone call by a click of a button the same way I do inside the network.

We have Alcatel Lucent Omni PCX office  when extension 221 is dialled  and if no one picks up, the call gets forwarded to another extension 283

I have admin access to the  portal  and not sure where the setting needs changed  and any help would be great.

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
I had this question after viewing Skype For Business Client Missing Call Forward Option.

I am unable to select the voicemail option when trying to forward using Skype Online

I'm trying to figure out the best way to connect our current phone system to CRM.
Right now we are rocking with a block of DID’s on a PRI and our phones run through the internet on a dedicated PRI from Comcast.
We have an NEC PBX connected to the Comcast service with NEX UX5000 phones for the sales team.
I don't know where I would start to find a financially viable option that replaces our setup.
I want to keep our current phone numbers, Would I replace out PBX with a cloud PBX such as ring central which easily integrates with CRM's such as salesforce?
Would I replace everything including the Comcast PRI trunk service and go with an all in one package?
I have little phone system knowledge I just want a phone system that can integrate with CRM and has access to answer calls on a phone and see calls come up on the screen.
I am trying to export a list of all Directory Numbers and their Description from CUCM into an excel spread sheet.  I don't need all  the fields,  just the description and extension
When we installed the AVST outlook client for voicemail it added IPM.Note.AppliedVoiceTechnology.VoiceMessage to every users Personal Forms library. This add in does not play nicely with Win 10 so we want to remove and have users listen to their voicemail messages (.wav files) with media player. When we uninstall AVST it does not remove the form from the Personal Forms library of the user, and having it there the user receives an error when attempting to listen to voicemails in Outlook. How can we remove the IPM.Note.AppliedVoiceTechnology.VoiceMessage from all users Personal Forms library's?
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?

I have Asterisk version 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.
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.

Software version 7.5.6a
Hardware version 1.0.4


No recent updates have been made.

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?
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.
I'm trying to use Twilio to create a ringless voicemail with Twilio.

I'm using PHP to write this in ideally.

I'm open to other options as well.

Can anyone tell me how I would go about doing this?
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?
I have a Mitel 3300 which I am connection to an AudioCodes SBC for teams direct routing. Making progress but stuck in two spots. One I make a call from teams to either an outside line or extension, the phone rings, but when you answer disconnects immediately. Looking at syslog it shows a 305 incompatible media format error. I cannot seem to find what to change on either the Mitel box or SBC to correct this. I tried filtering codecs in Mitel but no luck.
Server 1 receives the secondary ISP IP rather than the primary. Server 2 when I go to ipchicken and every machine on my network receives the IP from my main ISP. Server 1 is the only one when I do ipchicken shows the IP for ISP2
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
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 ???
We've setup a GETVPN between multiple sites.

It works fine for all traffic except video telephony.

CUCM to CUCM is ok for voice and video

CUCM to UCME is ok for voice and video

UCME to CUCM only work for voice. With Cisco CP-8845 when the call is placed, it rings at the CUCM site but when the user at CUCM site picks up the phone he gets silence, and video does not work
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

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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IP Telephony