Voice Over IP

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Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).  Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).

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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.
0
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I am trying to setup our client side pc's to use the pass-through Ethernet port on our Avaya 1608-i phones. I am having an issue that I can only connect one device to the port on the Cisco switch at a time. If I plug the phone in to the switch it works and if I plug the PC in to the switch that will work fine. If I use the pass-though port on the phone then only either the phone or the pc will pick up an IP.

We are not using a separate Vlan for VoIP.

is there a setting on the switch that I am missing?
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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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Several weeks ago I was surprised to get a Skype call from a relative who was traveling overseas. My surprise was not due to the fact that she was calling, but that Skype was running. I had not started Skype, nor have I ever configured Skype to start automatically at system startup! I thought it was just an anomaly, only to discover several weeks and at least as many reboots later that Skype was still indicating that I was online!!? When I check the usual places (the taskbar, etc.) I am not finding any indications that Skype is running. When I checked the services list, the only thing I am saying is the Skype updater. Does anyone have an idea what the heck is going on? I do not want Skype running unless I expressly permit it!!
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Hello all,
Here is the issue.
We have one user that uses a VoIP phone that is trying to go through our network.  He is a CxO and he is the only one that is using a VoIP phone.
He keeps saying that packets are dropping.
I just turned on 802.1p tagging but I do not know what else to do for a single phone.
Any help would be greatly appreciated.
Thanks,
Kelly W.
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I recall a few years ago that I used a program similar to Skype which allowed me to have private conversations via this tool.

There was some sort of key that I generated on my PC and emailed to the other person, which that person added to this tool. We then have "private" conversations.

Does this sound familiar to anyone? That does tool, or another, offer this today?

Thanks.
0
Hey guys,

We have a FreePBX system, how can i access the recordings?
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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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MyNetFone is a VoIP company that allows an SMS to be sent by running the following:

https://www.mynetfone.com.au/send-sms?username=[username]&password=[password]&to=[destination]&subscriptionId=[subscription id]&text=[message]  

Open in new window


Details on this page https://www.mynetfone.com.au/Residential/Home-Phone/MyNetFone-VoIP/features/MyText-SMS/API

What would be the best way to trigger the above code when receiving an email with a specific subject or from a certain sender?
Thanks for your help
0
  • 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.
0
How do I go about moving from a phone number hosted with Grasshopper.com to Google Voice?
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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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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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I stumbled on this 'free' service through a website that promotes deals on the web.

Figured I'd give it a try and it can't hurt to have a spare sim with live service but I have a bunch of questions and their tech support is the most useless I have ever experienced.

Wonder if others here have used this service and can play tech support for them since theirs functionally doesn't exist.
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could i see the
do not call list


how can individuals know which numbers not to call; if they cant see the list


I am not sure which zone this question should be in so please add zones.
0
Does anyone know why the Avaya Phone System would give an error "DHCP ACK" error bootup? I noticed that it seemed to happen during the day when it was busy and towards the end of the day, it worked fine. When it did occur, I ran a ping test and noticed that it had timed out so it led me to believe that it was probably related to network congestion.  Also, not sure if the IP Office phone system's port is running only at 100MB (and don't think that system can handle more). We will be monitoring this some more through out the week.
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On a network there are VoIP phones, the workstation are connected through the phones.

Is it possible to create a Vlan for the phones and a VLAN for the other devices?

I will enter the phones MAC address , and that list of MAC will have their own VLAN. all other devices will be on a different VLAN.

if yes how do I accomplish that?

( I want to use Sonicwall)
0
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I call 1800 numbers and press 0 many times and then get disconnected

and then I call back and press 1 many times


is there an automated solution to call 1800 number press (0001, 0002, 0003 , 0004... ) and alert me a human has picked up phone
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Hey Guys,

How do i backup the address book on a Cisco SPA525G phone?
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I have CUCM and Unity Connection at my main office.  We just acquired a small office in another state, and want to put them on our VOIP system at our Main office and have a couple questions:

1.  Will the telco allow me to port numbers over from one state to another (different area codes)?
2.  If they do allow that, how would I ensure those numbers we ported have there own opening greeting?  So if someone called those numbers we ported over - they want it to ring a separate open greeting.
0
If a home user subscribes to FIOS, do they need to purchase new phone equipment for the apartment (i.e. answering device, headset, conference phone, etc.)
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.
0
I have a Cisco Linksys SPA 8000.  I was wondering if there was a way to export and import the current config.
0

Voice Over IP

8K

Solutions

10

Articles & Videos

7K

Contributors

Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).  Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).