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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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Hello,

We are using FreePBX, and have Cisco 525G2 phones, and we added a SPA500s sidecar, how do i configure extensions on it?
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One of a set of tools we're offering as a way of saying thank you for being a part of the community.

Need 2-line analog phones (not VOIP).  Can't seem to find anything reasonably priced for my client.  Their needs are pretty simple:

1.  Telephones (quantity 4) for two-line analog.
2. Must be able to indicate when a call was missed or message was left, by way of an LED on the phone.
3. Must be able to page the other phones (hands free) by pressing a page button.
4. (optional) Intercom capability.

They currently have 2 of the Cortelco 7-series 2740 phones. These have been replaced by the 2750, and they're over $200 each and take 3 weeks to ship.

Anyone have a better idea for 2-line phones?

Thanks.
Dave
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We have a new issue on our phone system.  If we receive a call and try to transfer it to another internal number, but that person cannot take the call...if we go back to the original caller and try to transfer the call a 2nd time, transfer is not available.  This all of a sudden started happening.  We are on CM version 11.0.1.22900-14.  Thank you.
0
We have an HP 2920-48 port POE switch. We also have 2x HP 2920-48 port NON-POE switches. I was surprised to find today that our new VOIP phones are actually get power when plugged into the NON-POE switch?! Is this possible?? If so, are there possible issues I should consider when configured in this manner. I know we are having issues with the phones losing calls before they can be answered...wondering if related.
0
What is the process for adding a ten digit (area code + seven digit phone number) phone number to an Office 365 Skype for Business (Lync) account?
0
Do the Comcast Panasonic KX-TPA65 VOIP phones also have a network jack for plugging in a PC (in addition to plugging the phone into the network)?
0
Anyone know why my log buffer in my functioning voice gateway 2911 is full of this message and how I can stop it? debug is not turned on....:

SIP: Trying to parse unsupported attribute at media level

I see a bug report at https://quickview.cloudapps.cisco.com/quickview/bug/CSCuv36964 but they have no advice for how to stop it from constantly filling up my log buffer .... does anyone know how to make this irritating message stop? Is there any way to determine exactly what the "unsupported attribute" message might be that triggers this or where it is coming from? If someone has an old test sending out something repeatedly, am I going to have to sniff packets all day to find it?
0
What are the best types of 5 port POE switches that can have Comcast Power over Ethernet (POE) phones plugged into them?

An office where I work with have two Comcast POE phones but has only one Ethernet port. We need to connect a switch to it so we can plug in both of the Comcast POE phones as well as two laptops.

What are the best types of switches for this?
1
Hi,

We have a Polycom SoundPoint IP 350 and a 450 at a remote office that connect to a FreePBX 2.11/Asterisk 11.7 box in our main office. The two sites talk using a site-to-site VPN via our FortiGate 30E firewalls. When the two remote phones register across the VPN, everything works great and there are no problems. However, if the internet goes out at the remote site, which it does often, the remote phones will never try re-registering again. Even restarting the phones by power cycling them does not let them re-register again. The two things I've found that works is to either upgrade OR downgrade the firmware by 1 version, or to restart the firewall at the remote office.

I've updated the firmware on both Polycom phones to the latest versions, applied the latest firmware to both firewalls (we have other remote sites that do not have this issue) and made sure that SIP ALG and VOIP application control are disabled on both firewalls. I can consistently reproduce the issue by unplugging the modem (not the firewall) at the remote site and then plugging it back in. When internet comes back up, the phones will not be registered and will never try registering again until the firewall is restarted or firmware version is changed. I've also played around with registration expiration and timeout settings on the phones, but this doesn't seem to work, either.

I'm thinking this may be a FortiGate firewall issue, but it's strange that my other 3 remote sites (using the same …
0
Hi all,

We've got a client with 2 managed Cisco routers (not sure what model at this stage and we don't have control of these) that have a leased line into these with different providers and they have asked us to setup failover for them so we were going to get 2 more Cisco's and put these in HSRP on the LAN side behind the managed Ciscos with one of the line as the main line and the other as the backup/failover line but they have SIP coming into one of the lines so is there a way to set a WAN IP to "float" between the 2 Leased Lines to point the incoming SIP trunk to ? They want to avoid having to call the comms company to get them to change the SIP endpoint IP should one leased line fail.
Hope this makes sense.
Thanks
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I'm running IP office manager v8.1
for some reason the call forwarding has stopped working.

I have the system to forward calls at night by manually hitting the "Night forward" button, which is ext 298 and it is forwarded to an outside number.

when i hit forward and hit that number, the calls don't get forwarded and they're not ringing at office either. help?
0
Hi All,
 
We currently have Business Voice Edge VIOP from Comcast which is their proprietary voice platform. They have provisioned and require a 50Mb circuit over their fiber backbone to our office to services SLA for their voice platform. Thus far, as far as reliability, I have to say that we have had no real issues with call quality over the past year of usage.
 
Management has decided to move offices earlier than expected, and we overlooked Comcasts terms and conditions regarding portability of service to locations that do not currently have a Comcast fiber  backbone in their building – which the location we are moving to does not have Comcast fiber. They are also not willing to work with hhus to temporarily provision over another circuit. At this point, we have three options – ranked in order of preference, and I wanted to know if anyone has experience and any recommendations to help in making the right decision. Here are the scenarios:
 
Upgrade with Comcast to the new location and wait 6 months for them to build out their own fiber (includes city permits) to the new office.
One of two options in this cast to get service to our new office:
                                                               i.      Implement RingCentral month-to-month as a temporary VOIP platform while we wait. Forwarding temporary numbers to main numbers.
                                                             ii.      Implement a Ethernet Dedicated E-Line (point to point) between our …
0
Hi All,

We currently have Business Voice Edge VIOP from Comcast which is their proprietary voice platform. They have provisioned and require a 50Mb circuit over their fiber backbone to our office to services SLA for their voice platform. Thus far, as far as reliability, I have to say that we have had no real issues with call quality over the past year of usage.

Management has decided to move offices earlier than expected, and we overlooked Comcasts terms and conditions regarding portability of service to locations that do not currently have a Comcast fiber backbone in their building – which the location we are moving to does not have Comcast fiber. They are also not willing to work with us to temporarily provision over another circuit. At this point, we have three options – ranked in order of preference, and I wanted to know if anyone has experience and any recommendations to help in making the right decision. Here are the scenarios:

1.)      Upgrade with Comcast to the new location and wait 6 months for them to build out their own fiber (includes city permits) to the new office.
a.      One of two options in this cast to get service to our new office:
i.      Implement RingCentral month-to-month as a temporary VOIP platform while we wait. Forwarding temporary numbers to main numbers.
ii.      Implement a Ethernet Dedicated E-Line (point to point) between our office via layer 2 routing.
2.)      Terminate with Comcast and incur termination charges – something that we don’t want to do.…
0
We're running into call quality problems and I think part of the problem is that the VOIP real time packets are traversing many hops to the other side of the country before they finally make it into Microsoft/Skype network. My question: Is there any way to choose which gateways to use for peering with Skype for Business 365?
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I hear that waiting until the URL for a meeting arrives for first set up WebEx can be time consuming. So, is there a way to do this initialization beforehand?

Thanks.
0
I am attempting to force all ShoreTel IP Phones onto the voice VLAN.  However despite various attempts the connected devices remain on the default VLAN.  Due to computers connected to the pass-through Ethernet ports on the phones we cannot use the primary VLAN for the port.

I have a phone with a computer in pass-through connected to gi6.  I have been attempting to coax the switch to place the phone on VLAN 10.  Yet I consistently receive the following output:
switchb#sh mac add int ge6
Flags: I - Internal usage VLAN
Aging time is 300 sec

    Vlan          Mac Address         Port       Type
------------ --------------------- ---------- ----------
     1         00:10:49:45:8c:26      gi6      dynamic
     1         08:2e:5f:07:b1:7d      gi6      dynamic

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Running Config:
switchb#sh run
config-file-header
switchb
v1.4.8.6 / R800_NIK_1_4_202_008
CLI v1.0
set system mode switch

file SSD indicator encrypted
@
ssd-control-start
ssd config
ssd file passphrase control unrestricted
no ssd file integrity control
ssd-control-end 
!
vlan database
vlan 2,10,65,200-201
exit
voice vlan id 10
voice vlan state oui-enabled
voice vlan oui-table add 0001e3 Siemens_AG_phone________
voice vlan oui-table add 00036b Cisco_phone_____________
voice vlan oui-table add 00096e Avaya___________________
voice vlan oui-table add 000fe2 H3C_Aolynk______________
voice vlan oui-table add 001049 ShorTel
voice vlan oui-table add 0060b9 Philips_and_NEC_AG_phone
voice vlan oui-table 

Open in new window

0
got a missed calls from last two months, This evening I got a ring from +381 628302791 at 4:08.Whoever calling rings 1-2 times and disconnects immediately.
searched on google Which country code +381, It's from Serbia I shocked why I got calls from this area.
I would like to know, is this going to be a problem? Like hacking etc??
0
I have a subnet (10.201.x.x) used for VOIP. The .1 gateway resides on an HP 5400 series switch in order to provide multicast services to a number of internal networks across our WAN. We have just installed a Cisco Express Gateway for Jabber clients to utilize VOIP services from home, across the Internet. I set up a PBR on the voice vlan to use 10.201.0.3 for a gateway when calls come in across the Express Gateway and to use 10.201.0.1 for a gateway when communicating with our internal networks.

A key server with IP 10.201.0.5 has Trusts established for LDAP authentication to our Call managers. When the PBR is in place, the Trusts do not work for some internal domains because the .5 server is now routing to the Internet for DNS rather than looking to the local conditional forwarders to identify the domain controller of the internal network attempting to authenticate.

I know you can set up a PBR to be port specific but the Jabber client and Expressway use many tcp and udp ports to communicate. Is there a simple way to force the .5 server to resolve all DNS internally yet route all other traffic pertaining to a Jabber connection out the 10.201.0.3 gateway?

Here is a snippet of what is not working:

Policy pbe 10201-routing
     150 class ipv4 "nls-1916814"
      action ip next-hop 10.200.200.100
      exit
     160 class ipv4 "kms-172"
      action ip next-hop 10.200.200.100
      exit
     170 class ipv4 "10201-Internet"
      action ip next-hop 10.201.0.3
   

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0
I have a client with about 30 users that work at home about 50% of the time.  They continue to have connection issues logging into Citrix, RDS and now a Parallels server.  The problem is not what we are logging in to, which is what they think.  We have the users mostly hardwired to their router so we aren't dealing with wi-fi issues.  There are various internet vendors (Comcast, Verizon, ect) mostly in the DC area but some scattered across the eastern states.  Everyone works fine for awhile (like the last two weeks, barely a problem) but then we have various issues that turn out to be a user working from home, locking a file or getting disconnected repeatedly.  We might have 10 users logging in from DC office to Berkeley without any service disruptions during that same time.  

Before we start ripping servers out, I think we really have to make sure that the WAN is working correctly.  That starts with home internet.   Most of them are running a company provided VOIP phone at home.  It doesn't seem related but maybe it is.
0
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I have a Cisco WS-C4503 chassie with two WSX-4148-RJ45V 48 ports blades.  When I connect one of my NEC VOIP phones up to the system I do not get any power to the phone from the port.  I have check to ensure port POE is set to auto.  Here are some of my configs, I assume these blades are POE since they have DC In-line on them.
_________________________________________________________________________
Cisco_MFG#sh environment status
Power                                             Fan      Inline
Supply  Model No          Type       Status       Sensor   Status
------  ----------------  ---------  -----------  -------  -------
PS1     PWR-C45-2800AC    AC 2800W   good         good     good
PS2     none              --         --           --       --

Power supplies needed by system    : 1
Power supplies currently available : 1

Chassis Type : WS-C4503

Power consumed by backplane : 0 Watts

Supervisor Led Color : Green

Module  1 Status Led Color  : Green
Module  2 Status Led Color  : Green           PoE Led Color : Green
Module  3 Status Led Color  : Green           PoE Led Color : Green

Fantray : good

Power consumed by Fantray : 30 Watts
_________________________________________________________________________

Cisco_MFG#sh power inlin
Available:1400(w)  Used:0(w)  Remaining:1400(w)

Interface Admin  Oper            Power(Watts)     Device              Class
                            From PS    To Device
--------- ------ ---------- …
0
I see sub-domains available. What's it do?
0
What is the best practice of backing up Cisco Call Manager 11.5 ? (VM)
0
Digging into Call Manager 11.5

What is cube ??
0
We just installed ShoreTel phone system connected to a PRI with 20 DID numbers .
We also ported our main number (say 453 270 9999) from another provider to the same PRI line.
how do I route my older number which was ported to an hunt group or a single extension on a PRI?
when I dial  453 270 9999 from outside I am getting a prompt from Shoretel to dial an extension number.
I have to conigure it to ring to a hunt group or a single extension.
0
I installed WhatsApp and need to help someone connect to me. What do I provide?
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).