Hello I'm using softphones as my only 'landline' phone service for almost 3 years now (Diamondcard and now voip.ms), so far using SIP (and mostly Twinkle). Also, I'm using Linux (Debian) as my choice of desktop OS. Also, sometimes I'm in networks with badly behaving NAT routers (for some time I used openvpn to solve this unreliably, then I ended up using 3G instead of wifi while in Canada, but now I'm abroad and don't have 3G). I'm now sufficiently fed up with SIP to give IAX2 another try.
I want a softphone solution that: * works on Linux (Debian) * works reliably (e.g. remain connected for incoming calls, work with shitty NAT routers) * preferably encrypts both signalling and voice (dunno if voip.ms supports it, I might use a proxy asterisk instance on an own server instead) * properly handles audio with the 8000 samples/second dictated by the POTS systems (ALSA combined with some hardware (like both of my laptops) doesn't do proper lowpass filtering for mic input, so I will have to either use OSS or PulseAudio or rely on Asterisk doing proper downsampling in software). Asterisk seems to fit the first three; I'll happily build a GUI on top if this turns out to be a stable solution. My problems right now: - when I issue "console dial" without a number, it plays a recording with a woman's voice, and I can understand what is being said, but it sounds very garbled, like modulated with some about 20 Hz signal (a bit like a robot voice). What could be the problem? (Not using pulseaudio; +- default configuration.) One hypothesis I have is that it uses a too small buffer somewhere. - I don't understand how the extensions stuff is working. voip.ms wiki told me to create sections named [voipms], but how do I switch to 'default'? tie*CLI> console dial 4443 No such extension '4443' in context 'default' tie*CLI> console dial 04443 No such extension '04443' in context 'default' tie*CLI> console dial 004443 No such extension '004443' in context 'default' - I haven't found anyone in google who tried to do the same as me, except http://www.junghanns.net/en/asteriskassoftphone.html but that doesn't lead me far (and the patch linked is unavailabe). Has anyone here done what I envision, or seen some docs specifically matching my use case? Thanks Christian. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
