asterisk users - Dec 2004 - RE: Hook/Flash, Hold, Call Waiting, Three Way Calling

Do you have threewaycalling and transfer set in your zapata.conf?  Here's
mine (four TDM400's, seems to be working so far).  I didn't do anything
in
my extensions.conf for any of these features (what confused me at first is
the t and T options of the Dial application in extensions.conf are for
transfers via the # key), when you flash you get another dialtone that works
just like the first one.  I haven't tried three way calling yet, I only have
two hands, but transfer and hold (with music) work.

zaptel.conf
-----------
fxoks=1-11
fxsks=12-16
loadzone = us
defaultzone=us 

zapata.conf
-----------
[channels]
language = en
musiconhold = default

signalling = fxo_ks
context = internal
threewaycalling = yes
transfer = yes
group = 1
channel => 1-11

signalling = fxs_ks
context = pstn
group = 2
busydetect = yes
channel => 12-16

I've just picked up a pair of IAXy devices.  They work fine except that they
keep going offline.  As in, I plug it in, it connects to Asterisk, I can 
dial and phone and all is dandy.  Then, maybe 12h later, maybe 24, maybe 36, 
maybe 48, I'll either try to phone the device and not get through or
I'll
pick it up and the dialtone is gone.   it's simply lost it's connection
to
Asterisk.  If I unplug and plug back in, it reconnects and all is well.

I'm running firmware v. 22.

Anyone else experiencing this?

Paul

I've been messing with some hardware sip devices and with softphones 
(X-Lite, X-Pro and SjPhone).   Compared to the hardware devices, the 
softphones blow chunks (tm) for sound quality.  the softphones are quieter, 
crackly, and overal more difficult to understand the voice, while a sip 
device at the same location sounds great.

I've tried proprietary softphones (such as Nortel and one other who's
name I
forget) that sound great,  and X-Lite and sjphone are both using the same 
protocol and codecs as the sip devices, so i don't understand what I'm 
missing.

Anyone have any suggestions?

regards,

Paul