[Asterisk-bsd] chan_phone.so on Freebsd (Quicknet hardware)?

[msg]

Chris,

Thanks for your reply.

> I can't help very much apart from saying that the _r libs do not work. 

I rebuilt 1.0.2 (ports) with debugging and PROC defines:
	PROC=i686
	OPTIMIZE+=-O6
	DEBUG=-g
	OPTIONS+=-DLOW_MEMORY
	DEBUG_THREADS=-DDEBUG_THREADS -DDO_CRASH
	(MALLOC_DEBUG is broken, not used)
	
and '-pthreads' is used (no _r libs).

Behavior of the program has changed and no more bus errors so far:

(phone.conf is configured for dialtone mode, slinear or g723.1)
-------------------------------------------------------------------------------
Lifting the handset:
(no dialtone is heard)

Mar  7 14:16:49 WARNING[157144064]: chan_phone.c:874 do_monitor: Dial 
tone write error
         --- repeats until on hook or number dialed from keypad

-------------------------------------------------------------------------------
Dialing the phone from the console:

	-- phone rings
	-- lift handset, hear silence
	-- ringback continues on console
	-- after four rings on console, busy signal heard in phone
	-- console hangs up:
*CLI> dial 1265
*CLI>  << Console call has been answered >>
  << Hangup on console >>
-------------------------------------------------------------------------------

Dialing '#' on the phone, (slinear):

Mar  7 14:53:08 WARNING[155009024]: chan_phone.c:298 phone_setup: Failed 
to set codec to signed linear 16
Mar  7 14:53:09 WARNING[155009024]: chan_phone.c:566 phone_write: Unable 
to set 16-bit linear mode
Mar  7 14:53:09 WARNING[155009024]: file.c:550 ast_readaudio_callback: 
Failed to write frame

-------------------------------------------------------------------------------
Calling from the phone to the console (1234) -- g723.1 -- no dialtone:

	-- lift handset, silence is heard
	-- "Dial tone write error" messages scroll on console
	-- dial '1234', busy signal is heard in phone
	-- console messages:
<< Call placed to 'dsp' on console >>
  << Auto-answered >>
Mar  7 14:42:16 WARNING[157132800]: chan_phone.c:289 phone_setup: Failed 
to set codec to g723.1
Mar  7 14:42:16 WARNING[157132800]: chan_phone.c:543 phone_write: Unable 
to set G723.1 mode
  << Hangup on console >>

-------------------------------------------------------------------------------
Calling from the phone to the console (1234) -- slinear -- no dialtone:
	-- same phone behavior as previous
<< Call placed to 'dsp' on console >>
  << Auto-answered >>
Mar  7 14:46:53 WARNING[157132800]: chan_phone.c:298 phone_setup: Failed 
to set codec to signed linear 16
Mar  7 14:46:53 WARNING[157132800]: chan_phone.c:566 phone_write: Unable 
to set 16-bit linear mode
  << Hangup on console >>

-------------------------------------------------------------------------------
Configured SIP: added an extension aliased to 10 at nat1.cybertheque.net 
called '1235';

  when called from the console, works ok
  when called from attached phone (1265):

Mar  7 16:17:33 NOTICE[135398400]: channel.c:1691 ast_set_write_format: 
Unable to find a path from UNKN to SLINR
Mar  7 16:17:33 WARNING[135398400]: chan_phone.c:298 phone_setup: Failed 
to set codec to signed linear 16
Mar  7 16:17:33 WARNING[135398400]: chan_phone.c:566 phone_write: Unable 
to set 16-bit linear mode

-------------------------------------------------------------------------------
whenever chan_phone.so loaded:

         -- digits dialed from the console during a connection (IAX, SIP)
            fail to arrive at destination or arrive after long delay.
         -- without chan_phone.so loaded; with ixj.ko, zaptel.ko, 		 
        ztdummy.ko loaded: digits dialed work ok.

-------------------------------------------------------------------------------
when booting, get this error occassionally:

loader.c line 102 (fini_modlock): Error: attemp to destroy locked mutex 
'&modlock'.
loader.c line 223 (ast_load_resource): Error: '&modlock' was locked here.
loader.c line 102 (fini_modlock): Error destroying mutex: Device busy

-------------------------------------------------------------------------------



> You are much better off with  FreeBSD 5.3 than 4.x for asterisk.

There seemed to be a notable absence of discussions of the merits of
fbsd releases in the lists.  I chose 4.11 since it seemed that 5.x 
removes support for needed hardware, uses DEVFS (we need to statically 
configure devices) and requires DHCP for net booting (we need to use 
BOOTP) among other things.
> 
> I have been around the asterisk FreeBSD project for most of its life and 
> I don't recall anyone trying the quicknet products.
> 

I had hoped that previous posters to the original [Asterisk] list who
mentioned Quicknet hardware support in the fbsd ports were writing from
experience.

> I cheat and use ztdummy and a couple of sipura 3000 boxes. A number of 
> people use the xp100 real/clone cards and the 400 series cards.

Our intent is to build a small footprint netbooted asterisk box which
serves as a SIP ATA (avoiding the cost of Sipura 3000s) which will
support at least one fxo (PTSN) and two fxs (analog phones); we must use
the analog instruments (which have special features for disabilities)
so we can't substitute SIP phones for them.

I didn't see where linux would fit this requirement (am I wrong?)

Since I now have a starting point to debug the chan_phone.so problems,
is there anyone willing to join in?

Regards,

Michael Grigoni
Cybertheque Museum