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).

Here's the problem or issue I am having with Avaya Merlin Magix:

I have one lucent phone that is not working as the rest of the other phones or lines:

A. I must dial 4 digit extensions instead of 3.
B. I cannot retrieve my voice messages
C. I could make outbound calls instead of the phone calls go to front dest operator.

Please let me know how can I access VM and remove 4 digit extension and return 3digit ext?

Thank you
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My CEO travels to Hong Kong and China once every quarter. When he stays there for two weeks he usually uses his iPhone to make hundreds of minutes of phone calls via the Verizon roaming voice plan. That amounts to as high as $900 per trip. (The roaming rate is $1.99/minute in those areas.) That makes no sense. Can you recommend some cost-saving solutions?
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http://www.target.com/p/binatone-freetalk-starter-kit-for-skype-blue-talk-5365-e-r/-/A-14485138

I am using binatone usb headset for windows 7
I extend range with usb extension cable
I am not sure if my computer is usb2 or usb3
I am not sure if usb extension cable is usb2 or usb3
I am not sure if microphone, headset to usb is usb2 or usb3

Will I have issues if I get a longer usb extension cable
Will I have issues with usb2 vs usb3
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I have given up on portable usb/bluetooth headsets.
I do not need to walk far, maybe just 50 feet.
Please do not suggest portable headset. I have opened many questions and broke many devices and call quality is worse. Portable usb receiver stops finding my headset and I reset computer and then buy a new device. I have done this many times so please do not suggest portable headset for this question.

So I try to buy
http://www.ebay.com/itm/50-FT-Hi-Speed-480Mbp-USB-2-0-Extension-Cable-with-Active-Repeater-U2A1-A2-50-/271033783064?pt=US_USB_Cables_Hubs_Adapters&hash=item3f1adf5318

50 FT Hi-Speed 480Mbp USB 2.0 Extension Cable with Active Repeater(U2A1-A2-50)

what is Active Repeater?
will this make sound better for non portable usb headset

Is 50 ft the maximum
Does long extension cord affect call quality?
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Hello Experts,

I am in a discovery process and my company has a siemens PBX 9006 Model 80. I never did this before but I do have the login information to the box from the console. I want to prepare a document that will indicate whatever is necessary to remove the system and put a new voip solution.

From this question I just want the top level highlights of all the discovery points that I need.

Thank you.
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Hi,

Somebody at the office wants a rather strange setup for the Lync-Voip phone integration. She want to access her computer from home through a VPN solution so she can access Lync on her office computer. This way, she wants to appear like when she initiates a call from Lync, it would rather appear she's calling from her desk phone. First of all, can Lync be integrated with a VOIP phone this way? Second, can you get Lync calls through a VPN connection? Maybe I'm complicating things too much...The main thing is she wants to work from home and when calling clients, it would appear like she's calling from her desk.

Thank you and sorry for the long message.
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I am using a Cisco 2821 Router as an h.323 voice gateway router and CUCM 7.1.5 for call manager.  I recently changed telco providers and am seeing inbound calls on the caller-id come up differently than they did with the old provider.  For the most part all inbound calls show the name and number, which is how we want it, but we have a few customers that call in, and now since changing providers just their phone number shows up, not the name, but it did with the old provider.   So like I said call-id does work for the most part, but for certain customers that call it, it just stopped showing the name.  Its weird.  Would this be something on the telco side?  Her are my relevant caller-id configs:


isdn switch-type primary-ni
!
!
!
voice service voip
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 h323
  h225 display-ie ccm-compatible

interface Serial0/0/0:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-ni
 isdn incoming-voice voice
 isdn supp-service name calling
 isdn outgoing display-ie
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We received the following email from our SIP Trunk provider:

With the Cisco Call Manager w/ CUBE, the CUBE is not passing UPATE messages to the Call manager. Thus, the CUCM will disconnect the call after 30 minutes, since no UPDATE was received. The CUBE must remove the UPDATE in the "Allow-Header" field.

This command will remove the UPDATE from any request or response message type:

voice class sip-profiles 1
request ANY sip-header Allow-Header modify ", UPDATE" "" 
response ANY sip-header Allow-Header modify ", UPDATE" "" 

The "voice-class sip profiles 1" must be applied to the dial peer facing PAETEC.  In this example we are using dial-peer 9000

dial-peer voice 9000 voip
destination-pattern .T
voice-class codec 1
voice-class sip localhost  dns:astrial.pe.mcleodusa.net
voice-class sip outbound-proxy ipv4:xx.xx.xx.xx
voice-class sip early-offer forced
voice-class sip profiles 1
session protocol sipv2

So here are my questions:
1) We need to make these changes in the SRST device which is a which is a Cisco 2921/K9, correct?
2) The first part looks pretty straightforward, but what about the second set of instructions? I guess I need the Dial Peer explained to me and which IP address is supposed to be in the outbound proxy.
0
Can someone send me the step by step on how to create DHCP scope to handle out option 150 for Cisco Phones VOIP?
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I have one very large site 50 staff, with asterisk based SwitchVox PBX system, and one small office with 3 staff / 3 phones.

The sites are connected with comcast ethernet, traffic goes from local company switch through comcast connection to local company switch.

Analog call (small remote office B) -> audiocodes (small remote office B) -> ethernet switch (small remote office B) -> ethernet switch (large main office A) -> router (large main office A) -> PBX (large main office A).

At the PBX there is a rule that sends the traffic back to staff at the small distant office, so the entire connection looks like this.......

Analog call (small remote office B) -> audiocodes [ VLAN 1] (small remote office B) -> ethernet switch (small remote office B) [ VLAN 1] -> ethernet switch (large main office A) -> router (large main office A) -> PBX [ VLAN 2] (large main office A) -> router (large main office A) -> ethernet switch (large main office A) -> ethernet switch (small remote office B) [ VLAN 1] -> VOIP desk phone extension (small remote office B).

When the calls come in and reach the PBX, they are fine.  The quality is fine. The recording played by the PBX is crystal clear.  But when the calls get routed back over the same path to an individual extension at the small remote office and connect, the voices sound garbled.  The call is connecting, the caller and receiver of the call can hear and speak to each other.  But it's nearly unintelligible.

The PBX has definitive …
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So I have installed many on-premise Cisco VOIP solutions in the past, but was asked today by a customer who wanted us to host their VOIP in our datacenter since we already host many other applications for them.  They currently have an old PXS that really needs replaced, and they could really take advantage of some of the features of Cisco VOIP.  They have made it very clear they don't want this on site.  They have a large internet circuit and they do have a complete cisco infrastructure with QoS capabilities to ensure VOIP traffic will have priority.  The question I have is regarding the phone lines themselves.  They currently have a PRI T1 that their phone lines come in on at their location.  If I wanted to host their VOIP, I would have to to have all those numbers ported over to a PRI T1 in my datacenter right?  Or should the phone lines still remain onsite, but ip phones will just register over the internet?
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I would like to permit only users with a known PIN to be able to dial known heavy fraud countries like Turks and Caicos etc. Is there any means to effect this in Cisco Unified Communications Manager? My other option would be to use partitions and calling search spaces and limit access to certain phones or DNs. But PIN access would be even better.
0
Can I have all my phones and extensions traverse across one SIP trunk? Or do I need multiple trunks?
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Hi Experts,
Is it possible to dial outbound calls on my Cisco IP SPA502G, initiated from my PC.

I'm not after a softphone because phone needs to be functional when PC is turned off.

More like, after a cheapish software tool that tells the phone to dial xyz

Many thanks for any tips.
String
0
We are using a ESI 200 phone system.  WHen we try to call out certain places (BofA) the number keys don't make a sound, so we can't navigate their systems auto-attendant choices.

Any ideas?
0
Getting a number of users complaining of poor voice quality.
I have spoken to both our comms and phone suppliers, not much help.

i've been notified of issues at both our sites.  Both sites are configured the same (in terms of hardware) the phone system is shoretel.
The l3 switch going through a cisco asa firewall is an hp 2910 al poe.  Routing all traffic to the firewall.
The only different i can see is the remote site has not got QOS enabled to Differential Services.  SHould this be enabled in order to help jitter at site b?

What else can i check/enable to help with jitter.
Our 10mb links are running at about 2mb on average.
0
on windows 7
I can receive phone calls on skype

for months I can not hear most of the voicemails

Where are they stored and what is wrong with my skype configuration.
0
I can only make local calls from a pots phone plugged into an FXS port on a Cisco gateway. I can make long distance calls from our VoIP phones, which are Cisco and Cisco Call Manager. The dial-peers seem to be correct. Any help would be appreciated. Thanks!!
0
Can someone please provide me with a step-by-step guide on setting up SIP trunking in CUCM? Thanks
0
Hello All,

I have inherited a number of Asterisk servers running Asterisk SIP Phones in several offices at the company I am currently working with. I am currently running:

Asterisk 1.4.19.1 ON CentOS 4.4

As far as I know, there is no GUI front end and I can only configure anything from the CLI.

One of the offices is moving location and as part of this, they are also changing the phone numbers.

1- Do I need to make any changes inside Asterisk?
2- How and where?

I hasten to add, that I'm not a telephony engineer, so a lot of stuff is a bit foreign to me. I am a Windows engineer really :)

Please could anyone help?
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I have been using a CIsco 3845 for trunking SIP traffic to my PSTN carrier. It's been working fine in this role and it has dual power supply which i love.  And it can pass quite a bit of traffic as I have it doing some routing as well. Am I missing out on anything by not upgrading to the CUBE?
0
I am using CUCM 7.1.5 and I have two H.323 voice-gateway routers which are Cisco 1760's. The first router has two VIC2-4FXO cards with 8 analog phone lines plugged into it, and the second router has one VIC2-4FXO card with 2 analog lines plugged into it, which gives me a total of 10 phone lines. We are growing out of 10 phone lines, so I just ordered a voice PRI (T1). I will be activating this circuit next week. I am going to install the PRI on a Cisco 2821 Router, and the 10 analog lines that are plugged into the 1760 routers will be ported over to the PRI, and I will then decommission the 1760's. My question is can I add this Cisco 2821 router to CUCM as a gateway now, even though there is no PRI connected to it at the moment because activation isn't until next week? I want to do everything I can ahead of time to minimize downtime during cut over day. Thanks
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Hi,

I have been told it is possible in Lync server 2013 to set up call forwarding / Simring, both server side and client side.

So that if user X is calls, it will also always call a number set server side. But still allow them to set up their own call forward through the client.

Any one know how this is done.

For example I want it so that if you call user A, it will also always ring the reception phone. However user A can still forward his number to another phone such as his mobile if he wishes, But calls will still be forward to reception as well.

Cheers
0
Hi

Im looking at trying to get hold of some traffic stats off our hp switch that drives our 2x vlans for data and voice.

The console is pretty pants.  Any pointers?

Thanks
0
I have nothing running in the background and yet when I run SKYPE, I get static, funny noises and the Skype message says that I need to quit any file sharing applications.

What can I do so as to improve my Skype calls.  
I'm using the free service.
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).