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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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
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Expert Spotlight: Joe Anderson (DatabaseMX)
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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?
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The phone is Polycom VVX 350, provisioned by RingCentral.

Is it possible to somehow program the Polycom phone to produce a distinctive ring if a specific number calls.  A mobile phone can do this.  I wondered if there is any way that I can get this feature? If not Polycom, is there another brand of VOIP desk phone that has this ability?

Thanks.
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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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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!
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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.
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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?
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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.
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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?
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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?
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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.
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I'm working on an Allworx Server, I'm trying to get a Handset Template to become active.

Can anyone tell me how to get that marked as active?

HandsetTemplatesPage-1-.png
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Someone spoofing my number and calling another person with mortgage offerings and other services.  
The area codes and prefixes of my number and other recepient's number are always the same.  Last four digits are always different.  Both numbers are with AT&T carrier in US.

Anyone had experience how to deal with it?  
Any solution besides keep blocking them?  Can AT&T do something about it?
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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
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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
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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
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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
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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
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Exploring SharePoint 2016
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Skype for business 2015 Yealink T48S | Trusted Certificate
I need assistance with phones we recently purchased, all T48S 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.
firmware
Scenario:
Phones register correctly for as long as the 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.

securitypage

i did some googling and found this command but its only relevant for SKype for Business online
https://docs.microsoft.com/en-us/powershell/module/skype/set-csipphonepolicy?view=skype-ps

What is causing the phones to download the internal root domain CA certificate?
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cisco 7841 voip phones reboot themselves randomly when internally transferring an outside call.

I have a draytek 2820, SIP ALG disabled, unifi 24 port poe switch  -  

some phones on switch poe, some on PSUs so not power related

I have tried a 2820n and 2860 router

it seems disabling RSTP on the unifi switch cures the issue (not 100% sure yet, further testing tomorrow)- why is this?

Or any other ideas what this could be?

Thanks
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We have a FreePBX 13.0.196.2 system running Asterisk 13.19.1 and we're using Asternic Pro 2.2.6 for call reporting at a call center. The problem we've had since the system went in is that all of our users are in a queue. If a call comes in, all the phones in the queue ring. Person 1 answers the phone and after talking for 30 seconds or so, transfers (blind or attended makes no difference) to person 2. Person 2 talked for, say, 50 minutes. As soon as the caller or person 2 hangs up, all time associated with the call is attributed to person 1 and person 2 takes no time for the call in the reports. We've tried both blind transfer on the phone as well as just pressing ##extension during the call with the same results. Is there anything I can do to get call reporting to work properly within Asternic so that call times with transfers are accurate?

Thank you
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I’m looking for a simple answer to a simple question that it seems our Mitel vendor is having some difficulties answering: are Mitel MiVoice 250s formerly (Mitel 5000) networkable at layer 3 over a private VPN to other Mitel platforms, namely Mitel MiContact servers/Mitel virtual MCD, and Mitel 3300

Our environment consists of the following.
 
Data center:
·         Mitel border gateway
·         Mitel MiContact server
·         Mitel virtual MCD

Corp HQ:
·         Mitel 3300

35 branch locations:
·         MiVoice 250
·         No Mitel border gateways at the branches
 
 
We presently send SIP traffic directly to the 35 MiVoice 250s “over the top” to our broadband circuits which provide excellent call quality.  During primary broadband failure, we’d like to have calls failover to our 4G/LTE cellular service, which we tested, and work almost as well (in terms of call quality) and are not limited in number of calls.  The only way we can have automatic failure to our 4G/LTE backup is through the purchase of an Mitel border gateway (x 35 locations) and additional SIP licenses for the backup cellular route.
 
We’d like to send SIP calls for all 35 branch locations to our data center Mitel border gateway, and have the SIP calls distributed to each of the 35 locations over our private VPN.  For each of the 35 locations, all data traversing the private VPN fails over automatically between the broadband circuit and the 4G/LTE backup cellular route.  Routing our …
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Hello,

I am wondering if anyone knows what cordless units are compatible with the NEC SV8100 phone system.

I need to replace a couple of cordless handsets but cannot find any, even used. There are plenty of corded DTL-12D-1 BK TEL DT300 units available but I require cordless. Ebay has some base sets but no accompanying hand unit. I'd like to get a couple of more years out of the system which is still working fine.

Thanks!
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Dear Experts, we got this requirement for PABX configuration in a hotel/resort:

ext.png
A master extension will be represented for each villa, then several sub-extensions will represented for blocks in villa. Any incoming/outgoing call will be directed via master extension. Sub-extensions will be used to calling inside a villa (between blocks) only, when calling outside, master extension shall be used.

For example the screenshot is showing Room 202 and its 6 blocks. It will have 1 master ext (8202) and 6 sub-extensions for 6 blocks.

So can NEC and Alcaltel meet this requirement? And what are the things we need to consider? Many thanks!
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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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