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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How can I set my Skype default options to be that anyone can present/share files when Skype is opened? My current set up means that I have a Skype meeting and then when the person wants to share I have to change the settings and then re-commence the meeting? It's set so that only people from my organisation can share...thanks
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This is 100k sqaure feet, two storey, new manufacturing facility, on its stage of network/architect design.
We estimate that there will be 200 spots that need network access, for computers, VoIP phones, WiFi APs, smart TVs and security cameras.
In about 80 spots of the above, a VoIP phone and a PC will coexist.  These spots are the sitting spots or cubicles for engineers, managers and office staff.

The main question - should we run two network drops or one network drop in each of above 80 spots?
Option1: Two drops - One for VoIP phone, the other one for the computer.
Option2: One drop - VoIP phone and the computer will be daisy chained.

Not trying to over complicate the above main topic, we do have a few other questions as below in case you'd like to share some insights as well
1. Should we separate VoIP phones and computers into different VLANs? Why?
2. If we put VoIP phones on a separate VLAN, will the above Option2 still be doable?
3. Should we deploy CAT6A or CAT6 cables? 10G network is getting popular.
4. Should we run all cables directly from the end spots to the server room? Or, install some switches in the middle?
5. Any other thoughts? Or anything we should be aware of?

Thanks in advance!
1
It was ugly with Skype. Still haven't figured how how to add (or invite) external contacts to chat in Teams
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Dear Experts,

I have two FreePBX virtual machines distributed over two different data centers but both are for the same company, I am planning to use sip trunk on those two VMs and seeking to get them to work in failover/load balance mode.

I know that some sip trunk provider provide the capabilities of failover in case my primary Public IP is not responsive however I would like to extend the failover and load balance to the level of the VM as well.

Have anyone tried to do the load balance/failover method on a VM level between two datacenters ? How would VoIP traffic react in case of primary VM down? how about end points configuration ? Can I direct end points to a single FQDN where both IPs can resolve and if one of the VMs are down still the end point would get register and act like nothing is happening ?

I would appreciate any person's comment that have had an experience with such scenario.

Thank you
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we are looking for 800 phone no with api. meaning when someone call 800 phone no. we want api to be called immedately to www.myweb.com/phone no
then redirect to 310.222.2222

do you know which provider can. do it? just like something out of the box without too much coding involved
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Looking for a VoIP phone system.
This client currently has a PBX black box supporting over 50+ phones. They are renting 5 lines from a local teleco.
They are building a very modern 100K square foot facility and will be moving to there in the near future. I started researching a new phone system to replace the current one.
I am not a telephone guy, so I would like to hear your opinions. First of all, I guess VoIP is the way to go, correct? I am kind of an old school and did not trust VoIP at the beginning. But after using Skype calling long distance for years, I find it has better voice quality than the landline most of the time.
I heard of 3CX, a cloud based phone system with no need to run an in-house PBX hardware or software. If the PBX is virtually on the cloud, how can we connect the 5 rented lines from the teleco to the virtual PBX?    
What solution would you recommend based on your real world experience? Reliability, voice quality, easy management.

Thanks!!
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I have a Cisco 2811 ISR that appears to only have 64MB in flash and just 33MB available.
I only need encryption K9 for SSH access to the box and I need to be able to send/receive
IP SLA checks for VOIP RTP. Can someone recommend which image I need to in download
for this? I tried to download Enterprise and that's nearly the full 64MB or more.
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Hi,

We use Mitel 5212 IP Phones. we are trying to get them to work on a custom VLAN setup on a watchguard m500 firewall. We have created the custom vlan and the ip scope which works fine. I have mimicked the DHCP options from our windows based dhcp server, however this didn't work. On the DHCP windows based DHCP server the options are:

128 Mitel TFTP xxxx.xxxx.xxxx.xxxx
129 Mitel RTC xxxx.xxxx.xxxx.xxxx
130 Mitel IP Phone Identifier MITEL IP PHONE
132 VLAN for Mitel IP Phone 0x3
133 priority for Mitel IP Phone 0x6

On the firewall dhcp scop options 9All custom)
Code       Name                                 Type            Value
128         Mitel TFTP                           IP                  xxxxxx
129         Mitel RTC                            IP                    xxxxxx
130        Mitel IP Phone Identifier  Text              MITEL IP PHONE
132        VLAN for Mitel IP Phone   Hex              3
133        Priority for Mitel IP phone Hex             6

When the phone eventually boots it gets a crazy VLAN id. Any clues as to what I am issing, or a how to guide on getting the IP phones to work?

Cheers,
Paul
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Need education on 5 WAN IP block (same subnet) and the MPOE running up a fiber connection to the office suite.    We walked into this situation illustrated below.  There is one circuit coming into the suite.   The internet service installed a 200 megabit fiber connection at the MPOE.  A couple businesses want their own separate public WAN IPs running off of this one circuit.   There is currently a couple TP Link routers that we like to replace.   What device (switch?  what kind of switch?  Any problems using one switch over another one?) do we use between the biscuit (one ethernet port) and the multiple WANs on the Sonicwall? Here's what we summed up the ultimate game plan below...

Use a Sonicwall Tz 500(a model with at least x8 interfaces) and configure 2 additional interfaces as WAN ports - this would then give us 3. Each of these we can configure with their own static IP accordingly. Next we would configure a LAN interface for each company. Then we would use Policy Based Routing to move traffic from example: LAN 1 "Company A" to WAN 2. Sonicwall also provides QoS I believe which will support VOIP traffic through the routing.


Capture.PNG
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CUCM 10.5 SIP Trunking

    I have a two site Cisco Call Manager Phone System with one server at each site (FL and California).   I have had SIP trunking up and running in Florida for about a year.   We are in the process of migrating our PRI trunks in California to SIP trunks, but we are unable to complete the RTP (Voice) connections on the calls.   Every time we attempt a call the new sip trunk which is mapped through our CA Firewall, the Call Manager Server at that site advertises the RTP IP for the server in Florida.   Since these are not mapped through the other firewall, the call fails with no audio.   I cant seem to find a way to make the secondary call manager server advertise it's own IP address for the RTP instead of using the IP of the publisher.   The calls originate from the CA server, it is just the RTP that keeps requesting to send to the wrong server.   Any help on  how to force the subscriber to advertise it's own IP or how to change it would be greatly appreciated.   At wits end on this one.
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Here is diagram for Voip phone connection.    phone1 ---- SW1 ----- Nexus7K ------SW2 ------ phone2
We configure auto Qos at two switches (SW1 and SW2), both switches could be 3560 or 4500 etc . Do you think we have to configure auto Qos or some Qos at the interface of Nexus7K which are connected to SW1 and SW2? Thank you
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Hi
I had issues routing voice traffic across two same make/model switches, one as a core and the other as edge.   Managed to resolve this by tagging ports in the respective vlans on the switches, simple fix but not something im used to.  
However we are now trying to get two older HP/3com switches to do the same.

Switch Config overview:

HP1910 switch using default vlan1 across all ports.  Connected to:
HP2910 switch vlan1 interface (acts as core and routes to firewall).  Connects to:
HP/3com2952 switch vlan1 interface.  This will connect to:
A new HP2910 switch.  This will in turn re-route back to the HP1910 above to form chain topology rather than current loop.  I'm assuming this is good?

I am not using any trunks or LACP.

Part one will be to route the new HP switch back to the 2952, once working ill then connect this back to the 1910.

So far i have created a new vlan20 on the old 2952 and tagged the port thats connected to the new 2910.   Here is the config:


 
[3Com Baseline Switch]display current-configuration
#
 version 5.20 Release 1101P10
#
 sysname 3Com Baseline Switch
#
 super password level 3 simple
#
 domain default enable system
#
 telnet server enable
#
 ip ttl-expires enable
#
vlan 1
#
vlan 20
 description Voice
#
radius scheme system
#
domain system
 access-limit disable
 state active
 idle-cut disable
 self-service-url disable
#
user-group system
#
local-user admin
 password simple
 …
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We have a Polycom VVX D601 IP phone AND we have a VVX D60 cordless phone that is supposed to register with the base phone.  It's not working.

We recently brought in some new fiber to the dry cleaners and with the fiber, our provider supplied IP phones (Polycom).  The dry cleaners need a cordless phone so the girls can walk around looking for clothes while talking to customers, so, we purchased a VVX D601 base and the VX D60 cordless phone that is supposed to work with the D601.

The cordless base will not pull an IP on the network.  We have plugged in the D60 to the router and it will not DHCP.  If we take the phone to our other office, an office with IP phones and a Xircom PBX, it pulls an IP.  If we take the D60 cordless to other networks that do not have phones, it will not pull an IP.  

Now, you're going to ask why we don't plug it into the switch with our other IP phones and the reason is our provider brought in a Juniper switch for the phones and they assign IPs on a static basis.  If we plug the D60 cordless into the Juniper switch provided by our phone provider, it of course will not pull and IP and our phone provider said they will not turn pan DHCP for us.  

So, my question is, since the base pulls an IP on network 1, which has IP phones and a PBX, but the cordless base will NOT pull an IP on any other network, is that because it's a phone and not a PC?

That's my guess.  The cordless D60 pulls an IP when plugged into a network with an IP phone PBX, but…
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can I block all voip calls
I dont want to block one phone number at a time

I dont want calls from voip phone numbers because it is usually a scam
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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
0
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?
0
3cx.. moved VM from one host to another and set static MAC. still no ext and cannot create TCP connection to activation.3cx.com
0

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).