asterisk users - Jan 2012 - Using Asterisk as a softphone

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.

Hi,
one reason for having that robotic voice could be improper codecs others
include low CPU processing power, memory not free etc. I once had the same
kind of issue with VAD(voice activity detection) turned ON from my service
providers equipment so my asterisk was performing poorly with VAD. Asterisk
version and its codec play more important role.

Regards,
Sammy

On Tue, Jan 3, 2012 at 6:34 PM, Christian Jaeger <chrjae at gmail.com>
wrote:
> 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.
>
> --
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