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5

Skype Problem ON even when OFF: Type /showplaces to a chat box to show the devices you are logged on. Type /remotelogout to remotely log you off from the other devives. Type /showplaces again to check


3

As per the links you provided, you must change your status to "Offline" before closing the application. Until Skype changes this behavior, it seems like your only 1-step solution.


2

Modern asterisk works same with pci card or without. Thoose recomendation valid for linux kernel <2.6.18, that is not actual in modern distro.


2

When I looked up the specs of this device they say it supports DNS SRV under RFC:2782. You can find the specs here You can find additional information about DNS SRV at RFC2782. Here is some good information using DNS SRV with a SIP implementation:


2

SSH protocol run on top of TCP part of the TCP/IP stack. SIP or other type of voip generally runs on top of UDP protocol. One of the significant differences between TCP and UDP is that TCP (somewhat) guarantees the delivery of each packet. UDP on the other hand does not. It is fire-and-forget. The consequence of that if there's some network hiccup TCP will ...


2

NAT provides security by obscurity as a side-effect of how it works, not by design. If you are concerned about security, you should have a firewall or packet filter on your router, which is meant to provide security by design, whether or not NAT is in the mix. As far as what access this gives people external to your network, it depends on the vendor's ...


2

A SIP trunk is a direct connection between your organization and an Internet telephony service provider. SIP trunk enables you to extend VoIP telephony beyond your organization’s firewall without the need for an IP-PSTN gateway. This simpler configuration is easier and less expensive to design, operate, maintain, and upgrade. Here is a suggested [blog]: ...


2

No. The equipment needed to run both DSL and VoIP (the modem and VoIP box) require an outside power source to function. Unlike traditional pots line VoIP and DSL cannot use the existing DC current on a circuit. Some areas offer the capability of plugging in an ordinary handset to these lines to reach emergency services in the event of a power outage. If you ...


1

Assuming a direct line to your DSLAM, this might work. Still, it depends on the network operator to supply emergency power to the DSLAM, which probably requires quite a lot of energy. Long story short: VoIP will not be usable in case of a disaster.


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As there are many settings that could influence the quality of VoIP, I recommend comparing all settings with those listed in the article : How to configure your router for blazing speeds. You should especially disable all unnecessary WiFi modes and protocols, for example "b+g WiFi mode" if you only have "n" devices, or 5GHz if you only have 2.4 GHz devices. ...


1

I think we can only guess at the authors intent and target audience (and bear in mind that was written in over 5 years ago now). I'd posit that the issue is that SSH is designed to handle TCP streams rather then UDP. Although I have found reference to tunneling UDP over SSH it appears to be a more complex hack - probably not something ideal for a VOIP ...


1

I would view it the other way around: The first difference is in the handset or the "phone" itself. Is the handset designed specifically for IP telephony? If not, then you need voice-enabled routers. Otherwise, you need handsets and routers which are compatible with one another. POTS = plain old telephone system. If you take an old phone off of POTS ...


1

This is, of course, BS. Voice over IP is general term referring to everything from Skype to SIP. Indeed, IP Telephony, although probably a marketing term by Cisco, is more of a synonym.


1

Each country has it's own top level "regulator" that is responsible for telphone number management. In Germany it is the Bun­desnet­za­gen­tur is responsible for structuring the national numbering space, setting the conditions for allocating and using numbers, and processing applications for numbers and number blocks. In the UK it is Ofcom This ...


1

See in particular the last step below "Port forwarding- Configuring port forwarding on your router can be beneficial and resolve many NAT related one-way audio issues". You made need to port forward or port trigger (port triggering is better) the ports used for SIP Signalling and Audio RTP. As a list resort put the VoIP phone in a DMZ. Source Solve One ...


1

There is full documentation about ports availible. It is not clear why you think it not availble. Asterisk use port depend of technology/channel type used For sip protocol asterisk use port described in sip.conf(default 5060) and for rtp data use ports described in rtp.conf(default 10000-20000). Usually sip use only udp port. TCP-sip can also use tcp port. ...


1

Have you checked that skype isn't still running on your phone? It happens quite often on my Galaxy S3 where I will exit the app but it still runs in the background without the icon in the taskbar and no notifications. Usually when this happens I just go to the app info and force close it.



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