[NBLUG/talk] Linux Telephony - Asterisk PBX - GnoPhone
Mark Street
jet at sonic.net
Sat Apr 19 23:47:00 PDT 2003
Well, I sucessfully installed my new Digium telephony cards and Asterisk PBX
server on my office RH7.3 Linux box to make sure I could do it before I had
to do it ; ). I didn't have a problem with linphone and SIP on my LAN, I
could change codecs to see which one worked best. gsm codec doesn't sound
too bad, I recorded some greetings over the phone through Asterisk into gsm
and they sound pretty damn good. We shall see how Asterisk handles IAX and
SIP over the net. I have been pricing the Cisco ATA186 phones too.
This is fun!!!!
On Friday 11 April 2003 11:25, Scott Doty wrote:
> Well, having now successfully associated GnoPhone with an Asterisk PBX
> (thanks for the help on that, btw), I do have a comment about the idea...
>
> I don't see any way to get GnoPhone to change codecs. Seems stuck with
> that tu-berlin.de gsm codec -- the quality isn't what I would consider
> telephonic. (More like a bad digital cell phone connection -- there was
> also some lag, reminiscent of talking over a satellite.)
>
> You might want to go with something else that supports g.711, like
> linphone, which is rumored to work with Asterisk. However, I haven't had
> much success getting linphone to work with Asterisk PBX -- indeed, my
> current config segfaults, apparently just after registering with the PBX.
> (I've already emailed their "bugs" contact, no word yet.)
>
> Meanwhile, the SIP phones (which are using g.711) have no distortion, lag,
> or jitter. They sound like an ordinary phone call. The Cisco ATA186 goes
> for $160 and supports two POTS lines. The cost of the terminals, of
> course, are offset by the cost of the PBX -- and I definitely think
> Asterisk PBX is the way to go. It is quite flexible.
--
Mark Street, D.C.
Red Hat Certified Engineer
Cert# 807302251406074
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