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4

I just got this working in Iceweasel 38.3.0 on Jessie. Mozilla says that 'Firefox Hello' is not available in ESR versions of Firefox but it seems that it's just disabled by default. To enable Hello go to the 'about:config' page and set the preference called 'loop.enabled' to 'true'. 'Loop' was a previous name for the 'Hello' feature. It seems that you also ...


3

If you want to hook it up to your phone line and run it locally you would need something to act as a PABX - In addition to the phones you will need a box which has an FXO port (to connect to the phone line), and PABX functionality - probably something running Asterisk PBX. (You could have the FXO port separate or built into the device). That said, that ...


3

Regarding architecture, considering the options in each of the two dimensions separately: Secured tunnel vs secure connection: In a VoIP scenario, usually there are many connections involved. VoIP clients register with a VoIP server, but for the actual call, the clients directly connect. Establishing a secured connection involves a huge overhead as ...


2

You cannot find the information because you, or possibly some optical character recognition process, have mis-spelled them. They are IAX (the inter-Asterisk exchange protocol, designed for communication between Asterisk systems as the name suggests) and IAX2 (version 2).


2

Usually a VoIP phone has two ethernet RJ-45 ports. One of them connects to the existing LAN using a cable that goes to the female RJ-45 on the wall, let's call it "External" and the other one using another cable that connects to your PC, that we are going to call "Internal". The VoIP phone has an Ethernet switch inside, this switch allows the traffic ...


2

On my iMac using Yosemite 10.10.3: Open System Preferences application. Select Sharing. Set the window as shown below. Plug device in to ethernet port. BTW, the iMac acts like a router.


2

Set up your home connection as a VPN then you will always be on your isp when you connect to the VPN.


2

Yes, it will be transparent, ie the phone acts like a 2 port switch.


1

A SIP trunk is only required to connect to the PSTN, it is not required for the SIP clients to communicate with each other. Your real issue would most likely be routing SIP through your firewall, most SIP servers require a Session Border Controller of some kind if the server sits behind a firewall or router, especially if NAT is involved. Assuming you can ...


1

One way to do it: Put a device, for example a Raspberry Pi running Asterisk. The fixed line can be connected to the DECT base and to an ATA with FXO port like this, to convert it to VoIP. The mobile phone will use wifi and a softphone as Zoiper or Media5-fone to connect to Asterisk using SIP. When you receive a call you can make it ring at the same ...


1

If you connect everything, and your phone audio breaks up or you start dropping calls, then you need QoS. It's not strictly correlated with number of machines on your network - you could have 2 machines on a network and if one machine is coming close to maxing out the network, and the other system is a phone, you'll need QoS. It's possible you could do QoS ...


1

Any host address in a subnet can be used as the gateway address as long as you assign it to the router. The router is really just another host on the network.


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Simpleast idea - setup openvpn for example with network 10.0.0.0/24 and pool 10.0.0.10-100 After that just reload asterisk and add to sip.conf localnet=10.0.0.0/255.255.255.0 All clients have connect to 10.0.0.1 Please note, you have start asterisk AFTER openvpn


1

DTMF are dual-tone analogic tones, analogic phones can send/transport/receive them easily because the tones are in a frequency range that phone systems are capable to handle. VoIP systems are different, they use codecs to convert voice to data packages. Some of the codecs don't compress audio but most of them do. When you take some audio (usually human ...


1

No, a VPN is not at all necessary. In a lower security set up, you can simply supply users with an extension, a DID, and a password, and they'll be able to make and receive calls from any internet connection (including their mobile phones while connected via wifi). Keep in mind, you'll need to purchase service and DID (Direct Inward Dial) #'s from a SIP ...


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I've solved this finally for me. The problem is, Telekom Germany does not like, when video codecs are advertised. After I've disabled every video codecs for this SIP connection, I am able to make a call outbound and not receiving error 606 Not Acceptable!


1

Your computer's noise reduction feature (assuming it has one) may be incorrectly telling Skype to cut out your outgoind sound. I am on a Mac, and have what seems to be the same problem, so here is what I am trying: I did the "Echo" test to hear a recording of a call, and I could hear my voice cutting out. I went in to System Preferences -> Sound -> Input ...


1

Which one is more secure and efficient for VoIP: PPTP, L2TP, or TLS... Here is some general information related to the question... VoIP can use dynamic compression codes and codecs, so they can be insecure regardless of the underlying encryption technology. See, for example, Spot me if you can: Uncovering spoken phrases in encrypted VoIP conversations. ...


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What you can't do You can't put pcm_mulaw audio or H.263 video in MP4 container. The error you provided by itself is generally not informative, but it is always accompanied by the actual error (at least for ffmpeg, not sure about avconv). Probably something this: Could not find tag for codec h263 (or pcm_mulaw) in stream #0, codec not currently supported ...



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