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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On my mac I use iMessage to send text messages.

What is there on Windows to do that same thing?

Thanks.
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What is SQL Server and how does it work?
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What is SQL Server and how does it work?

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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.
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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?

Thanks!
1
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.
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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.

http://www.cisco.com/c/en/us/td/docs/unified_computing/ucs/c/hw/C200M1/install/c200M1/replace.html#wp1053068
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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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Hello

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?
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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?
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  • 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.
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How do I go about moving from a phone number hosted with Grasshopper.com to Google Voice?
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Microsoft Certification Exam 74-409
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Microsoft Certification Exam 74-409

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Is there a portal for AT&T SIP trunk customers? We've asked our AT&T sales rep and Tech consultant over and over for this information but they can't seem to find it. I want to be able to forward DIDs which are part of our SIP trunks. So simple - yet so complex for the behemoth.
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I have a Zultys MX250 and today it kicked everyone out and said password was reset logoff and log back in.

this is strange.. i am and Admin and i cannot get in to the device .. help!
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Hello,

Can anyone point me in the right direction? There are a few google hits for this problem, so it seems like a known issue.

///////Summary: On outbound calls, we are sending out caller ID, and it is showing as "restricted" on HD enabled mobile phones only. Older mobile phones and land lines display our caller ID correctly. The telco PSTN providers have all pointed to our call manager as not providing the correct 1TU-T E.164 standard. This is occurring on all 5 MGCP gateway PRI's across multiple local PSTN providers.


///////System Parameters
We have 2 call managers in our HA cluster, 1 publisher, 1 subscriber.
Cisco Unified CM Administration

System version: 10.0.1.12900-2

VMware Installation: 2 vCPU Intel(R) Xeon(R) CPU E5-2609 0 @ 2.40GHz, disk 1: 80Gbytes, 4096Mbytes RAM, Partitions aligned


///////Troubleshooting Steps
Here is the support forum posting of the same issue, I have performed the changes advised in this posting with no success:
https://supportforums.cisco.com/discussion/12746556/debug-caller-id

1) Under Service Parameters - Clusterwide Parameters - Calling Party Number Screening Indicator

Set this value to Callmanager Provides Calling Number (No success)


2) I have performed the changed to Call routing information - Outbound calls, tested with Calling Party IE type as both "national/ISDN", Calling Numbering plan "ISDN". No success.

Here are our test call DNs appearing in the debug isdnq931:
May  4 11:44:02: ISDN …
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Hi

i have a Zultys Zip 57i phone that a user is having a problem with working from home.  It says no network available,

how do i get this phone to work on a home network ?
0
Hi There,

A potential new customer asked me to do some technical work to some technical work for them as they're moving to the new office. Part of the scope of the project is to run data and voice cable. I'm fine with the data aspect of it since it's something i've done for years, the question is with the voice aspect of it.

They say they have a Hybrid PBX from Verizon where they have 9 land line and it gets split into 23 (lines?), i have not done any work with Verizon hybrid pbx but i managed another network where we had a hybrid pbx from fonality with 4 land lines and the server served 20 extensions,  all phones connections were regular data runs to the voice switch the the server connect to.

My question is this: based on what they told me (the Verizon hybrid pbx) the cabling for the phone system is simply regular "data" cabling of they need to be terminated to a 66 block?

P.S, Attached is a picturecloset picture showing a Nortel and Meridial box (i'm assuming the pbx) and tons of cables terminating on a 66 block.

I greatly appreciate your help.
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Hey Guys,

How do i backup the address book on a Cisco SPA525G phone?
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Hello,

I have installed a CUCM Publisher in a UCS server, I have another UCS server and I need to install a subscriber, What do I have to configure in a subscriber?
The licenses are shared between CUCM Publisher and subscriber?

thanks
0
I am running Call Manager 9.1.2 and Unity Connection 9.1.2 in my main office.  We have a small branch office, we ported their phone numbers over to the main office about a year back, and their Cisco IP phones now register over a VPN to our main office.  This works great until there is a WAN outage, then they are without phones.  So I was reading about SRST.  So lets say I get 4 analog lines installed at this branch office and install an SRST router (2911).   This will allow them to call outbound during a WAN failure, but what about inbound calls?  Their main numbers are not going to work because their main numbers were ported over to our main office.  So how will they still get their inbound calls?  What do people do in this situation?  I feel like it has to be common, but i can't seem to figure it out.
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Hey Guys,

We have a Cisco SPA525G2 phone here, it's just stuck on the CISCO screen even after reboots. Does anyone have a fix for this?
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Hi Experts,

We just implemented a new VoIP telephone system and we are experiencing so random cracking and static on some of the phone calls.


The implementation guys are saying this is a network issue but I do not think this is the case since we did a voice readiness test before we started the implementation phase and the report did not show any issues.


Any suggestions on what ma be causing his issue?

Any suggestions on a tool that I may use to monitor and track down any phone quality issues?

Thanks
0
We had an 'issue' with one of our VMs, upon reboot the Linux file system needed repair.

At around the same time I see - "Lost access to volume 5351714a-0e19f12c-cc94-d072dca029b6 (datastore1) due to connectivity issues. Recovery attempt is in progress and outcome will be reported shortly."


in the ESXi Host's Events  which seems to indicate a disk issue -


Successfully restored access to volume 5351714a-0e19f12c-cc94-d072dca029b6 (datastore1) following connectivity issues.	info	27/03/2017 10:01:05 AM		localhost.thefirstgroup.com.au
Lost access to volume 5351714a-0e19f12c-cc94-d072dca029b6 (datastore1) due to connectivity issues. Recovery attempt is in progress and outcome will be reported shortly.	info	27/03/2017 9:58:06 AM		datastore1
Successfully restored access to volume 5351714a-0e19f12c-cc94-d072dca029b6 (datastore1) following connectivity issues.	info	27/03/2017 9:46:11 AM		localhost.thefirstgroup.com.au
Logging to storage has failed.  Logs are no longer being stored locally on this host.	error	27/03/2017 9:40:46 AM		localhost.thefirstgroup.com.au
Lost access to volume 5351714a-0e19f12c-cc94-d072dca029b6 (datastore1) due to connectivity issues. Recovery attempt is in progress and outcome will be reported shortly.	info	27/03/2017 9:39:50 AM		datastore1

Open in new window


Also, the Health Status shows alerts storage alerts -> Storage alert
I understand that a heartbeat was missed to cause the "loss access to volume" and it was restored - but why? Wht does this indicate? and pls explain what I'm seeing in the Storage summary? Why does it shod "DriveFauly" on all and Alert on a couple? What else can be done to identify the cause?
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Hi,

I have a question for Skype For Business.

I would like to force an update, where my Response group on my client is updated (Forced).

Here is my workflow :

I create a new distribution group
Then i hide the distribution group in the AD

Then i add members inside the Dist. group.


Now i go to the skype servers webinterface. https://skype4business/cscp

I create a new Group and under "Agents" i select : Use an existing email distribution list"

Group-S4B.png

How can i now force this setting out to my client, so they dont have to wait for ?! 8 hours !? for this change - and my TestWorkFlow would appear on the clients?

ResponseGroup-S4B.png
Hope this makes sense :)

Thank you in advice
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I've been having a sporadic issue with our 3CX VoIP server. Occasionally, we will get dropped incoming calls. I've been running wireshark across the interface and I'm seeing a ICMP Destination Unreachable, Port Unreachable. This is for WAN To LAN, LAN to WAN, and LAN to LAN. The one consistency is that it always involves the IP address of our 3CX server as either the source or destination.  This only started happening once we switched out routers. The system is running on Windows 2016 Datacenter in a Hyper-V environment.

Any ideas?
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I have a sound bite (< 30 seconds) that I want to send to my wife for her to use as a ring tone on her iphone.  We don't ever sync to itunes.

I have it as an mp3.  I used itunes on my win pc to make an .m4a of it.  I renamed it to .m4r.  Now i have 3 versions of the same sound bite.

Now what? I can email mail the .m4r or other file to her. (I have an iphone so I can test the process myself)

Love for it to be easy to do / no apps needed.. the harder / more steps, the less she'll want to deal with this : )

are there ios apps for this?
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Hi. We have a client using Snom 300 SIP phones (firmware version snom300-SIP 8.4.32) and he is trying to lock down his phones so that users can only call numbers in the phones speed dial list or an emergency number (UK 999). His speed-dial list is a mix of local, national and mobile numbers.
We can't control this through the telephone system as it is on a hosted service that doesn't have this functionality.  
Scratching my head over the dial plan really so, if someone is able to help offer a solution, this would be awesome.
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IP Telephony

6K

Solutions

9

Articles & Videos

5K

Contributors

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.