[Asterisk-Users] Re: Asterisk on SuSE 9.1?

[Rick Green]

thanks, Don.

I got past that hurdle last night.  I've been using the asterisk-update.sh
script, in conjunction with reading the asterisk doc project book, and the
quickstart guide on onlamp.com.

  While trying to simply find the zttool.c source and figure out how to
compile it separately, I discovered the other source trees, and while
reading their Makefiles, I learned that it did a 'uname -r' and used the
result as part of the path to the kernel modules directory.  Well, I had
run YOU, which updated my kernel and kernel-source, but I had not yet
rebooted, so 'uname-r' returned 2.6.5-7.95, but YOU had already cleaned up
that directory and replaced it with 2.6.5-7.111, so the linker was getting
a 'directory not found' error and reporting it to me as 'no kernel-source
found'.  I rebooted, then re-attempted 'asterisk-update.sh compile' and it
went to completion.  'make samples' and 'asterisk -cvvv', a bunch of
messages fly past too fast to read, and I got a CLI> prompt!

I browsed the sample configs from another xterm, discovered the 'console
extension', so I plugged in a headset and dialed the digium demo server.
It wasn't intuitively obvious to me, so it took several tries before I
realized that I could issue other 'dial' commands to send digits to the
IVR.  The audio was quite choppy, with about 2-3 'ticks' per second
superimposed on what would otherwise be clear audio.
  I then tried calling my own extension, which expectedly sent me straight
to voicemail.  I left myself a message.
  I then called the voicemail number, entered the mailbox number and
password (I thought that when you called voicemail from within the system,
it assumed the CID extension as the desired mailbox, so I was surprised
that it gave me the first prompt.)  I played back my message, and got
nothing but silence!
  ...so I apparently have a problem with one-way audio.  I know my
microphone is good, because the microphone audio is looped to the speakers
as I talk.

<non-asterisk-specific rant>

 (WHy is this?! I've noticed it with every sound card I've ever tried, and
it infuriates me that I have to deal with feedback from the analog
loopback in the sound card!  Supposedly these soundards are full-duplex,
so why are they looped by default, instead of keeping the inputs and
outputs totally separated!)
  ANybody know how to do an alsa.conf or set a mixer to fix this?

</non-asterisk-specific rant>

 ...so that's how far I got before climbing into bed at 2AM.  Once I get
audio transmit working so I can record and play back a message in the
voicemail app,  I'll define other extensions as the kphone, iaxphone, and
ohphone installations on some of my other boxes, and see if I can call
between them.  Once that works, I'll break down and spend some real money
to rent a DID and start shopping for an ATA and/or IPPhone.

-- 
Rick Green

"They that can give up essential liberty to obtain a little
 temporary safety, deserve neither liberty nor safety."
                                  -Benjamin Franklin