Voice Over IP

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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When booking a Skype enabled room that a Polycom Trio isnsigned in as, what are the expectations and controls over who can walk up to the phone and join the meeting?
It seems like a privacy/security issue.  
If you join the call from the conference room phone, how can you identify yourself as the organizer? Is the only way to do that to join from another device signed in as your organizer account??
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I only want to use classic skype for windows 10

please dont recommend another product or another skype

very difficult to reach dial pad

when I am on hold, I am asked numerous times, do you want to keep on hold, press 1

and I need to click 5 places to see dial pad
Is there a shortcut
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We are looking at moving from an in house phone system to a cloud based VOIP system. The issue is we only have one Ethernet jack at each workstation.  I have heard that it is not a good idea to connect the ip phone to the jack and then connect the workstation/laptop to the ip phone's Ethernet jack.  Does anyone have any experience with this?  Does it work or does it slow down the workstation, cause connectivity issues, etc.?  Any comments are appreciated.


Thanks,
cja
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Hi

We are adding a 4th switch to our network.  Not sure how best to configure.

Currently (see attached (new in red)) we have 3 connected switches:

HP 2910 L3 as core with two older v1910 and 3com 2952 switches connected via cat5.   The core switch routes traffic out to firewall.
The 2910 also has two vlans configured for data and voice.  The older switches are data only.

The new 4th switch will be in another office (c10-20m run away) and is a 2910al POE also.  We need to hook up poe phones here and desktops on vlans 20 and 1 respectively.

Main questions are:
  • Do we need two cable runs from the main 3 switches to the 4th?  i..e switch 1 to 4 and switch 3 to 4 (chain mode instead of looped?)
  • If two, can we mix the connections used to connect the switches i.e. fibre and cat5 or do they need to be the same throughout the switches e.g. cat5 only?
And if two connections do they have to be routing between the vlans e.g. vlan1 connected to vlan1 on switch 1 and 4 and vlan20 between switch 1 and 4?  Or does iprouting resolve this?

Thanks
Network.jpg
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Hi All,

We currently have an issue with a new build at a remote site.

The overall voice network is fully working at other locations, however the new site is having issues with inbound calls from the PSTN. The phones at both ends (internal and external) will ring, however no audio is passed. The call remains open, but silent.

Calls work outbound from the site successfully. The CUCM/Cube are on the main site, where calls work fine. The remote site is connected to the main network over a site to site VPN.

The only difference between this and other sites is the allocated IP range. The Cisco phones on the remote site are all using public IP addresses, where the main network and other remote sites are utilising private address space.

Any thoughts or suggestions would be greatly recieved.

Many Thanks,

John
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What is the difference between a Cisco phone "factory reset" and a "wipe" of the phone.

The code to begin the "factory reset" reset is: While the phone is powering up, and before the Speaker button flashes on and off, press and hold #. ...
Release # and press 123456789*0#

The code for the wipe is:3491672850*#

What is the difference?

Also, I have a Cisco 7960 phone that will not clear IP addresses and other settings for either process.

Any ideas how do completely clear the Cisco 7960?
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Hi

Does anyone know what the state is with the Shoretel (mitel) director licences?  Particularly with extensions?  We need to add more and cannot find out much info about them.
Awaiting comms co to update me.
Same question regarding the phones.  I believe 212k are EoL.  We need the line keys so what options do we have?

Thanks
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NEC SL2100 phone system started behaving poorly a week ago Sunday.  Rebooted all network equipment, and nothing changed.  Reboot the phone system and everything worked for about 20 minutes.  Same cycle of events, about same time frame.  Next checked all network equipment for firmware upgrades.  Wireless bridge Engenius EnStationAC went from 2.x to 3.1.2.11 on both sides successfully.  Problem returned yet again, but now a delay in getting dial tone of about 3-10 seconds.  If phone hangs up and connects immediately, no delay happens.  A monitoring computer is the only non-phone device on the entire voice network.  It shows random packet loss to phones.  The phone problem was originally they begain restarting at random.  They are not all happening at one time.  The delay never happens to VOIP phone on the system side of wireless bridge.  Engenius has yet to comment on the issue.  One phone (Only one phone) peforms IGMP actions accoring to Wire Shark.  Removing that specific phone solved the problem for almost 2 hours.  Yet it eventually returned.  All the time I have brought another laptop in for testing and moving it to both sides of the bridge.  If I am on the system side, I perform ping tests to the phones on the far side of bridge.  If I am testing the phone system resources, I am on the opposite side of bridge.  I perform around 1,500 packets to 3 devices simultaneously each time I test.  No latency or packet loss ever.  I have introduced a new PoE switch on the system …
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Dear Experts, is this diagram correct?

voip.PNG
We have 500 users, intend to use both VoIP service and Analog Tel service for backup. THere are 17 analog numbers to keep, so we choose Gateway GSM1024; for VoIP we use Grandstream UCM6510. We have several questions, can you please suggest?

- Can we use both UCM6510 and GSM1024 at the same time?
- Can we configure so that 1 department will use Analog line (GSM1024); other departments will use VoIP (UCM6510)?
- If one of these 2 fails , can the IP phone automatically change to the other, so that telephone service will NOT be interrupted?  
- Where can we find the reference link and installation manual for these devices?

Many thanks!
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Wiping Cisco phones with the code: 3491672850*#.

How do I confirme that the phones are indeed wiped?
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3cx.. moved VM from one host to another and set static MAC. still no ext and cannot create TCP connection to activation.3cx.com
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Is the Jabra Pro 9450 Duo Stereo headset fully compatible with VOIP services like WebEx?
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I have a Cisco 2960x switch and a heap of Nec DT800 handsets and would like to put them on a Voice VLAN. The only way I have been able to get it working was to enable the Voice VLAN on Switchport and configure VLAN tagging on the handset as we may need to use the built in switch.

Considering they move handsets often is it a safe assumption to just apply the Voice VLAN on all the switchports just in case they move one to an unconfigured port?

What’s the best practice?
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I want a cheap 800 number service. 3 choices with prompts. All I want is voip. There is no call center. No forwarding to cell phones. Maybe just used for voicemail. A free 1 month trial and an expensive bill is expensive.
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Hello and Good Morning Everyone,

          I recently added another phone line to my AT&T setup.  The hardware hookups are as follows:  The telephone/copier/fax machine has a direct telephone line connection to the back of the AT&T Gateway at the port labeled Phone Lines 1 & 2.  Then, I have another telephone line running from the Ext port of the  telephone/copier/fax machine  to the chordless phone.  That said, an AT&T agent explained to me that the  telephone/copier/fax machine with the direct connection to the Gateway will get the new activated number and the other phone, chordless one, will retain the old VoIP number.   Given this information, could someone explain what is meant by VoIP or Voice Over IP?  I did not want to bog down my AT&T agent with too many questions.  So, I decided to submit this one for review here.

            Any shared thoughts and explanations in simple terms with respect to the definition and function of VoIP will be greatly appreciated.

            Thank you

            George
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Difference between VOIP and SIP.  Could someone give me a practical description of what the main differences between VOIP and SIP are?
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This is about the switch infrastructure using Cisco switches. Currently, there is only using one Cisco WS-2960x-48 POE switch. We also using Cisco UCS 500 series for the VOIP. We are using vlan 101 for data, and 102 for voice. Please see the attached cisco switch configuration.

Now, we intend to buy one new Cisco Meraki MS120-24 ports switch, and join this switch into the switch infrastructure. We also intend to add-in 2 more VLANs for our new VMware virtualization management and backup segments. This is a new 2-hosts virtualization (vmware), with 2 network ports to form a trunk carrying existing vlan 101 (data), management (vlan 121), and backup (vlan 122) from each host. How should I update in my existing POE switch and also the new Meraki switch? Can I make all the 3 vlans - 101, 121, and 122 routable but only allow selective ip to access. For example, only allow 192.x.x.25 to access all vlan 121 & 122 only, but not the other way round.

Thanks in advance.
Cisco-2960-48-POE-Switch.txt
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Shoretel and switch STP on/off?

Hi

Looking at replacing our switches from procurve to Aruba.   Changing the method from daisy chained via ports to a stacked method using same models.  

Im unsure if we need to have STP disabled for shoretel to function?  If this is the case we cannot stack, which i find odd.

Thanks
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Ok, this is the last part to getting our site paging worked out.
I need to know how I can get our ShoreTel phones speakerphones to work with Valcom speakers when a page is sent out.
We have a ShoreTel system 14.2 director and Valcom v-2003A paging controller.  We need our system to page all phones and all Valcom speakers at the same time.
Can this be accomplished?
If this scenario cannot be accomplished with our current equipment then what would I need to replace/upgrade?
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Hello everyone,
We have a ShoreTel VoIP phone system and we would like to use it to page different zones from the ShoreTel desk sets.
We have a 3 zone Valcom Page Control Unit.
Our paging goal is this:
  1. Page outside only
  2. Page inside only
  3. Page outside and inside at the same time
Is there a solution to accomplish these scenarios?
Do we replace our current Valcom 3 zone paging control unit with a new paging control unit capable of accomplishing the 3 scenarios?
Thanks
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What options exist for porting a Google Voice phone number over to other services?

A couple of users who I support are interested in doing this and want to find out what other services they can port (transfer) their existing Google voice phone numbers too.
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I have installed Google voice on my Sprint PCS iPhone X.

Now how can I make my Google Voice number replace my Sprint PCS iPhone X caller ID number so that everytime that I call someone my Google Voice (and not my regular Sprint PCS phone number) will display on all outgoing calls?

Where do I go to enable this setting?

I received the email below on June 1, 2018 and I'm not sure how this affects my Sprint PCS Google voice service:

We are contacting you to let you know that per previous communication, due to upcoming upgrades to Sprint’s network, Sprint will no longer be supporting the Google Voice with Sprint integration. Effective today your Google Voice integration with your Sprint phone number (xxx) xxx-xxxx has been disabled by Sprint.

Effective today:
All outgoing calls (including international calls) and texts will be made through Sprint at Sprint’s calling and texting rates, if applicable.
All new messages, calls, and voicemails sent from your Sprint phone will not be stored in Google Voice. You will still be able to see your messages, voicemail, and call history from before June 1, 2018 in Google Voice on your Sprint device. You can also export this data from your Google Voice account at takeout.google.com.
You won’t be able to use Google Voice-enabled capabilities such as call forwarding, voicemail transcription, spam detection, and other Google Voice features. These capabilities can be enabled from your Sprint device. Click here for more information …
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I have a remote extension (yealink handset) that is dropping outbound calls after 35 seconds. I have diagnosed through logging that the asterisk server is not receiving the proper ACK, and the reason for that is that the asterisk server is looking for the reply from 192.168.1.4, which is the local subnet of the remote phone. Obviously the asterisk server will never receive a reply from that address. What changes do I need to make so that the asterisk server is looking for replies from the WAN IP of the remote network? NAT is setup properly on both ends. For reference, this is CompletePBX from Xorcom that we're dealing with and a copy of the asterisk log error is below.

[2018-08-16 09:27:08] WARNING[5720] chan_sip.c: Retransmission timeout reached on transmission 405286438@192.168.1.4 for seqno 2 (Critical Response)
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I am running the newest version of Skype on my Windows 10 PC.  I need to contact someone (I have their Skype name), but when I try to connect I get the following message:

Skype Error
I don't use Skype often, and tried to find a setting or other evidence of the problem, but can't.  Can you help?

Thanks,

Phil
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I am converting to SIP at a company site.  I set up the fortigate 60d identical to another one of our company sites where SIP is already implemented.

The SIP Cloud provider tells me that the firewall is not "passing confirmation packets  the phone system (PBX) sends out when it detects an incoming call",
and this is causing their system to transfer the call to our failover number.

I have opened up all requested ports (TCP/UDP etc), configurd policy,and QOS to high priority- everything they suggested, and as I mentioned earlier, it is configured exactly like our other site's fw.

I suspect it is a configuration with one of their servers.  In any case, Logging for all events is enabled in the firewall policy.  How can I tell if the firewall is blocking/not passing back the confirmation packets they SIp provider mentions?
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Voice Over IP

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).