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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).

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We have a shoretel system that is not under warranty support anymore that we have been asked to do some hopefully minimal support on as part of our day-to-day IT support for a client.

the client has 4 different locations with shoretel IP420/IP480 phones.
I am trying to move a phone from one site to another.....and I can not get the phone to show up in the list of available phones at the new site.
the phone has a static IP on the proper LAN - no VLAN.....i went over the phone menu settings with a working phone to make sure all the settings are correct on the phone.
the phone tries to download a configuration from the server - but says it failed downloading

what am I missing?  maybe licensing?
are there log files on the server side I can look at to see what failed?

as always - any help appreciated!
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I have to participate in an interview. I rather not but I have to do what I have to do.
Does anyone have any question on (hard and easy)
1.LAN
2.WAN
3.VoIP
4.file servers
5.Email servers
6.patch management
7.Support high-volume group printers, Xerox printers
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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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VG248 needs to be replaced. Anyone can tell basic step or send a link? Thank you
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I'm tearing my hair out on this one!

I've got a client who is having a VOIP issue (ongoing for several months).  Here are the facts:

1. Very small client: 4 users on a local domain, Windows PCs and a Server 2008R2.
2. VOIP phone system using same router as computer network.
3. Two phones work solidly.  Two are intermittent.  Tried varieties of plugging in phones, including directly into LAN ports on router instead of the network switch.
4. Intermittent symptoms are: 50% of the time a caller calls the number and gets either a fast busy signal, or "your call cannot be completed as dialed".  Sometimes the caller goes directly to voicemail even though the phone is not in use.
5. One of the phones has been replaced, as an experiment. Used to be a SNOM 320, now an Aastra model.
6. On the working phones, unplugging the phone on the network results in the caller getting voicemail.
7. On the intermittent phones, unplugging the phone typically results in the caller getting a fast busy signal (even at times when the phone seems to be working).
8. At a time when a phone was not working, I ran a continuous ping test to the phone from a computer on the LAN, and it never dropped out once.
9. Before the problems occurred, they enjoyed excellent service on this LAN hardware for some number of years.

The VOIP company is saying that the problem is either the switch or the router, though there are never any detectable problems on the network (there is a Windows 2008R2 server …
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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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In Skype for Business 2015, are there any APIs that I can use to query if a user has Enterprise Voice enabled?

I need to be able to differentiate users that have calling capabilities vs users that do not. Maybe being able to query if a user has a phone number associated with them would suffice too.
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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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Dear Experts, I have a question related to telephony service. We are using IP PBX Grandstream UCM6510 with SIP trunking from The Provider.

So as my understanding, for example if our number is +AA 710xxxxx; I create a conference room in UCM6510 at ext 8888; then when customers want to join a conference room with us, they will call to +AA 710xxxxx, press 8888. Am I right? (AA is my country code)

But the Boss now have some customers in USA, UK,... and he wants his customers will call to USA, UK numbers, (for example: +1xxxxxxxxx; +44yyyyyyyy) respectively instead of our number (+AA 710xxxxx)  to join our conference room.

Is this feasible? Can you please suggest the solution? Many thanks!
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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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Dear Wizards, is there any tool for simulating Grandstream UCM6510?

just like in Network we have GNS3, EVE-NG, Packet tracer,,...Can you please suggest? Many thanks!
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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 have one DC  with DHCP and DNS all in one. I am trying to connect a phone but it does not get an IP from the DHCP, Rebooted the server still getting (The DHCP service failed to see a directory server for authorization) error.

The phone (Cisco IP phone SPA 504G)  just sits on utilization network.
All other devices get IP and the lease time is set to 1 day.  It is when I try and add a new phone.
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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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Hi guys,

I just got a new Helpsdesk user. I want to add his extension to the existing ring group in SV9100 nec phone systems.
Could some one help me with this, as I am very new to phone systems
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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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Hi all,
I'm currently facing a strange problem.
I have two Freepbx servers located in different locations.
Location 1 has a sip trunk for incoming and outgoing calls and everything works perfectly.
The second location don't have a sip trunk yet and we're going to implement one soon.
But the local communication is working too.

Both sides are connected with VPN site to site.

I can register a phone in site number 1 into site number 2 PBX and it works.

I can also register a phone in site number 2 into site 1 PBX and it's also works.

My problem is that I was trying to connect both PBX with trunk beetween them, but no matter which trunk I've used (IAX2 or SIP) when I'm trying to make a call from one site to the other site I get "extension is unavailable".
I've checked and the trunk is up.

On both sites I did outbound route (on the first site he is located before the trunk to the sip provider) but still can't get this to work.

Any ideas?
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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).