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6

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

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 ...


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 ...


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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!


2

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


2

If headset has specific software install latest version, and update headset firmware. With Jabra headsets you can install Jabra Direct, update firmware, and it resolves these issues (most times).


2

Maybe check out this type of approach. I'm a little confused over it, but I'm looking for something similar and this is what I came across. http://accessories.us.dell.com/sna/productdetail.aspx?c=us&l=en&s=&cs=04&sku=A7791506&dgc=ST&cid=293344&lid=5616479&acd=12309152537461010&ven1=sE1inYhPj&ven2=,


2

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


1

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 ...


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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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.


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

Best option - contact onsip.com support. Auth username used ONLY if you want use for authentification other(hidden) name. Since you know nothing about that, leave it blank or put value of username in it(both option will work same). Password have be put in secret field.


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You have to connect the SIP:Wise to a VoIP/SIP Telecom provider, who will be your link to the PSTN (Landlines) world. Check with the main service providers in your country if they provide the service. However, there are international providers that give very good long distance rates. I use www.cheapvoip.com, but there are a lot of providers. Check this to ...


1

I have an answer, but it depends on how the Siemens box works. A quick Google search implied that it might be a digital device, which means that this suggested solution would NOT work. I did set up something similar to this for a client in the states, whose door intercom system was basically set up to make an analog phone in their apartment (that was ...


1

A VPN is your best bet. It is possible for providers to try and analyse what is going through the VPN by looking at the packet sizes and timing of transmissions, but this would only help them make a guess, and is very likely to be implemented. (If it were common, I'm sure that OpenVPN would have introduced functionality to allow stuffing packets and ...


1

To directly answer your question: Does Ventrilo use sufficient encryption to prevent eavesdropping while the connection is going through the internet? No, Ventrilo's encryption methods cannot prevent a knowledgeable individual from eavesdropping on your communications to and from the server. If you read further into the Wireshark article you linked as ...


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

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

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

I understand that you want that your Asterisk PBX use a dial-up modem as a PSTN analog card in order to make and receive calls, your Asterisk PBX convert it to SIP and put the call on a softphone. The big problem is that Asterisk is unable to "talk" to the modem card because there are hundreds of different models and Asterisk doesn't have the drivers for ...


1

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

The big thing about Asterisk is that you can configure almost anything to make it adapt to your needs. As your Asterisk PBX will be behind a VPN Server, you have to keep in mind the following considerations: When a client registers to the PBX, it uses SIP. By default SIP is being transported over UDP on port 5060. It can be changed to TCP on the file ...


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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Make sure to get one single codec in the 200 OK answer. Some servers will send you a second re-invite if they receive 200 OK with multiple codec in the SDP. To solve this problem, have a look at the current OK in the logs. Extract one of the codec from the list (your preferred one). Then allow only that codec in your Gigaset device. This should fix the ...



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