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  • how to start with a voip/public announcement project?

    - by Metiu
    The main requirements are: open source solution on Linux support P2P VoIP calls support presence support multicast VoIP announcements (and maybe some way of setting up such a "conference") preferably serverless (maybe the network can get split and I'd need to keep the functionality for all clients that still see each other) I tried looking at telepathy, in particular telepathy-salut, but it seems to be quite a new technology, so it lacks clear/good documentation and/or working examples. I'm also evaluating SIP (e.g. SofiaSIP), but it's working only if connected to a server.

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  • VoIP Phone Service - For The Every Communicator

    If you have already come across the term VoIP, but you are still not aware what does it mean what it has to offer you. You will be wondering whether it is beneficial to switch over from the old tradi... [Author: Dennis Smith - Computers and Internet - April 22, 2010]

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  • VoIP Phone Service For The New Era Of Communication

    The influence of VoIP phone service and its technology of Voice termination have grown over the years. This technology has facilitated the global extinctions of phone calls at low cost rate. There is... [Author: Dennis Smith - Computers and Internet - March 30, 2010]

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  • Get Connected Through VoIP Phone Service

    VoIP phone service technology is completely based on digital systems. This is certainly a remarkable way to stay connected at the cheapest rates with the people who are living at long distant places ... [Author: Dennis Smith - Computers and Internet - March 22, 2010]

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  • Is there a cheaper alternative to Skype for VoIP to PSTN calls to Vietnam?

    - by Nick Bolton
    My dad uses Skype to make calls to Vietnam PSTN, but finds that the rates are a little on the pricey side. It's probably not relevant, but he's living in Thailand right now. Is there an application similar to Skype, which is cheaper for calling Vietnam? The answer may well be no, since maybe the international telecom peers in Vietnam are just generally expensive... Who knows? While I'm asking, maybe it's worth someone mentioning if there's a cheaper alternative to Skype in general? I'm thinking that maybe not as Skype's pretty cheap anyway, but it's worth mentioning.

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  • Does anyone know of a good .Net VoIP library.

    - by Vaelen
    I have been searching for ages now for a good .Net based VoIP library. After having tried conaito and SIP.Net I have still haven't found anything that truly fits my needs. Basically all of these are constructed using ActiveX. Unfortunately ActiveX and WPF don't play well together. And ClickOnce Deployment doesn't register Interop COM components. I need to find a good, pure .Net managed code implementation.

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  • How does one capture H.323 voice traffic on a VOIP network?

    - by Chris Holmes
    What I am trying to do is capture the WAV data of a phone conversation on a VOIP network using SharpPCap/PCap.Net. We are using the H.323 recommendation and my understanding is that voice data is located in the RTP packets. However, there is no way to heuristically determine if a UDP packet is a RTP packet, so we have to do more work before we can capture the data. The H.323 recommendation apparently uses a lot of traffic on specific TCP ports to negotiate the call before the WAV data is sent via RTP. However, I am having very little luck determining what data is actually sent on those TCP ports, when it is sent, what the packets look like, how to handle it, etc. If anyone has any information on how to go about this I'd really appreciate it. My Google-Fu seems to be failing me on this one.

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  • HELP! need to make a VoIP program for WiMAX in NS-2 work..

    - by janiemack
    I've been trying to get this working for the past 2 days and i'm completely stuck. I'm trying to make a simple VoIP program work in the NIST module for WiMAX http ://community.4gdeveloper.com/attachments/download/14/090720150504_ns2-Release-2.6.tar.gz version 2.6 , using NS-2.31 (remove space btw 'p' and ':' ) http ://downloads.sourceforge.net/nsnam/ns-allinone-2.31.tar.gz?modtime=1173548159&big_mirror=0 (remove space btw 'p' and ':' ) The installation process goes fine. When i run this program, I'm getting an error saying " OFDMAPhy : error did not find match for permutation and bw combination" Some help would be really appreciated. thanks!

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  • Bluetooth Audio and SoftPhone Audio Input/Output

    - by o7th Web Design
    I have a Voip Softphone software that I would like to start using on my Ubuntu 14.04 box. Here's the thing. My system sound right now goes through my HDMI to my speaker system so I can play music all day ;-) I have a bluetooth headset connected to the machine as well. What I am wondering is if there is a way to: Auto-mute the music when a call comes in Auto-switch the sound devices when a call comes in, from my hdmi sound device, to my headset Auto-switch back when the call ends, and auto-un-mute the music Or even just an auto-switch to the headset? I can always pause the music ;)

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  • Default Critique branch office setup: VPNTunnel->HQ, subnets for VOIP/PC, + several Q's

    - by CHickenTaragon
    We're setting up a new branch office. * ~10 users. * Each user has a VOIP phone provided by a hosted solution. * Users need access to resources on HQ (located in another state), so setting up VPN tunnel * HQ only supports certain Cisco/Juniper devices. VOIP provider only supports SonicWall, so current plan is to have two routers w/ separate subnets for VOIP vs. PC traffic. * PC's will plug into pass-thru Ethernet jacks on the VOIP phones, but the phones vs. PC's will point to different subnets. * Cable Modem is 50Mbps / 5Mbps DOCSIS 3.0 business line w/ 5 static IP's. * Each of the 2 subnets will map to one of the 5 public IP's. * May or may not also need to support a VPN tunnel with a second branch office because of a file server they have there that some in the new office use. I'm pushing to have them move the files to a server on the HQ's network so we don't have to worry about setting up an additional tunnel. Questions: Do you foresee any issues with the below set-up? Router recommendations by HQ IT staff: Cisco Router 2811, or Juniper SSG5 or SSG20. Any recommendations about these routers? We need Wi-Fi too – looks like the above routers have models that support this, any reason not to use this? Users need to be able to work from home. If so, how is authentication handled? Right now we use AD credentials for the HQ's domain, but we currently don't plan to have an AD system in the new location since it's only 10 users. We can't tie the authentication system from the new location's router to the AD system of the HQ. All the PC's that will be in the new location are currently in the existing office that is closing down, and are already joined to the domain of the HQ. Please confirm: this + the VPN tunnel will be sufficient for them to connect to authenticated resources on the HQ's network from the new location, correct? Mainly SQL servers and file servers, and a few remote desktop sessions. I'm sure I'll have some more questions, but can't think of them right now.

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  • Finding cause of TCP retransmission within a LAN

    - by Surreal
    Hello denizens of Server Fault I have an irritating problem with a LAN of about 100 computers, 2 Windows domain servers, and 12 VoIP phones. Since their installation around a year ago, every week or so, we notice a VoIP phone resetting itself - occasionally in the middle of a call. Simultaneously there are often signs of temporary loss of connection on computers: freezes in explorer while accessing network shares, errors in our administration software due to loss of connection to the database server. I have been doing some Wireshark monitoring on the connection between the VoIP PBX and the rest of the network. Wireshark picks up a clump of retransmitted TCP packets at the times when we record phone restarts. The Wireshark log shows about 2 clusters of retransmissions a day ranging from 5 packets to hundreds. Those in each cluster are mainly between the PBX and some set of the VoIP phones, but not always the same set. Often retransmissions at the same time are to phones connected to the same switch, but sometimes retransmissions occur together to phones at opposite ends of the network. There are usually some coincident retransmissions in passing TCP traffic, for example between client machines and the file servers. The spikes in retransmissions and phone resets do not correlate well with when the network is heavily loaded. They seem to occur slightly more during the day, but most in the evening, when traffic should be decreasing. They occur reasonably often late at night when most computers are turned off and traffic should be lowest. Do you have any ideas that might help diagnose the cause of problems like this? One thing I have not yet tried, but should have, is updating the firmware of all the switches.

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  • Finding cause of TCP retransmission within a LAN

    - by Surreal
    Hello denizens of Server Fault I have an irritating problem with a LAN of about 100 computers, 2 Windows domain servers, and 12 VoIP phones. Since their installation around a year ago, every week or so, we notice a VoIP phone resetting itself - occasionally in the middle of a call. Simultaneously there are often signs of temporary loss of connection on computers: freezes in explorer while accessing network shares, errors in our administration software due to loss of connection to the database server. I have been doing some Wireshark monitoring on the connection between the VoIP PBX and the rest of the network. Wireshark picks up a clump of retransmitted TCP packets at the times when we record phone restarts. The Wireshark log shows about 2 clusters of retransmissions a day ranging from 5 packets to hundreds. Those in each cluster are mainly between the PBX and some set of the VoIP phones, but not always the same set. Often retransmissions at the same time are to phones connected to the same switch, but sometimes retransmissions occur together to phones at opposite ends of the network. There are usually some coincident retransmissions in passing TCP traffic, for example between client machines and the file servers. The spikes in retransmissions and phone resets do not correlate well with when the network is heavily loaded. They seem to occur slightly more during the day, but most in the evening, when traffic should be decreasing. They occur reasonably often late at night when most computers are turned off and traffic should be lowest. Do you have any ideas that might help diagnose the cause of problems like this? One thing I have not yet tried, but should have, is updating the firmware of all the switches.

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  • Using Google Voice with an internal SIP Server

    - by BHelman
    Let me be upfront and say first that I am new to the entire details of VoIP. My former understanding was just the extent of Skype. Don't worry, I understand a lot more of it now. The situation is this. I have a Google number that is actually very close to the area in which I live. It's convenient as it is not long distance for everyone. I love its features and etc, but I want it to forward to a VoIP phone, which will be my residential phone. Obviously, Google does not allow forwarding calls to domains (yet). So I use SIPGate with a SIPGate number to forward to a softphone for now. I can configure a VoIP phone to interact with my account easily enough. The problem lies with SIPGate itself really. Google Voice gives free unlimited inbound and outbound calling. SIPGate charges you for outbound. So a VoIP phone would work, but I could never make a call on it (for free). So let's say I setup an Asterisk server, or any other SIP server. What is the best way to go about linking my server to Google Voice? I looked into IPKall but it only specifies inbound calling and not outbound. Or is that just assumed? Does an SIP server handle outbound calling by itself?

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