asterisk users - Mar 2005 - Zapping around

Dear list,

I am trying to learn how to use Zap-things in Asterisk.

While loading Asterisk verbosely I get this error:

[chan_zap.so]Warning, flexibel rate not heavily tested!
 => (Zapata Telephony w/PRI)
  == Parsing '/etc/asterisk/zapata.conf': Found
Mar 12 17:19:01 WARNING[5563]: chan_zap.c:763 zt_open: Unable to open '/
dev/zap/channel': No such file or directory
Mar 12 17:19:01 ERROR[5563]: chan_zap.c:6208 mkintf: Unable to open
channel 1: No such file or directory
here = 0, tmp->channel = 1, channel = 1
Mar 12 17:19:01 ERROR[5563]: chan_zap.c:9155 setup_zap: Unable to
register channel '1-2'
Mar 12 17:19:01 WARNING[5563]: loader.c:345 ast_load_resource:
chan_zap.so: load_module failed, returning -1
  == Unregistered channel type 'Tor'
  == Unregistered channel type 'Zap'
Mar 12 17:19:01 WARNING[5563]: loader.c:440 load_modules: Loading module
chan_zap.so failed!
aaberga@epsilon-stargate:~> Ouch ... error while writing audio data: :
Broken pipe


Now, modprobing zaptel is ok:

aaberga@epsilon-stargate:~> sudo /sbin/modprobe zaptel
aaberga@epsilon-stargate:~> 

Same with wcfxo, wcfxs

aaberga@epsilon-stargate:~> sudo /sbin/modprobe wcfxo
aaberga@epsilon-stargate:~> sudo /sbin/modprobe wcfxs
aaberga@epsilon-stargate:~> 

although here I do have a first question. The card I am using is a TDM400
with two FXS module, an empty slot and an FXO module.

The wiki mentions a wctdm module that I do not find (modprobing it just
fails). Am I missing something, or can I use the older set of kernel modules?

[info: I did get the 1.0.6 zaptel, libpri and asterisk archives from the
Digium site; I did compile everything under SUSE 9.2, thus with a stock
2.6.8-24-default kernel;
I did use the make linux26 command in the install process of zaptel.
]

Now the card seems to react to my fiddling: the three green leds
corresponding to the installed module positions do turn on as soon as I
type the wcfxo or wcfxs modprobe command.

The zaptel config file is as follows:

************** zaptel.conf **************
the stock file as generated by the compile process, with the addition of
these lines


# edited by aaberga % 12.3.05
loadzone = us
defaultzone = us

fxsks = 1-2
#fxoks = 4

************** zaptel.conf **************


The zapata.conf file is as follows:

************** zapata.conf **************
the stock file as generated by the compile process, with the addition of
these lines


; edited by aaberga % 12.3.05

;signalling = fxs_ks
;context = incoming
;channel => 4

signalling = fxo_ks
context=internal
channel => 1-2

************** zapata.conf **************

I am obviously missing and/or misdoingsomething; can anybody help?

Thanks in advance
Aldo

Hi All,

I've been trying for a while to get * to play MusicOnHold with my SIP
connection. I can hear it when I call a test extension from my local X-Lite
phone, but when I dial in via InterVivo, I just hear silence.

I have a Gentoo box with kernel 2.4.28-gentoo. I have no sound card or speakers
on the box, it's in a cupboard.

I have uncommented the lines in musiconhold.conf. I am trying to use the
following extensions

[inbound-calls]
exten => s,1,Dial(SIP/07XXXXXXXXX,20,m)

[voip]
exten => test,1,Goto(inbound-calls,s,1)


Dialing test from X-Lite works correctly, and dialling in diverts to the mobile,
but with silence. PlayBack() with GSM files works okay.

Is there something special I need to do with InterVivo to get it to work?

Thanks
Jamie

SIP.CONF attached:

[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind SIP channel to
context = inbound-calls         ; Default context for incoming calls
srvlookup = yes         ; Enable DNS SRV lookups on outbound calls

disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=ilbc

externip = XXXX
register => 0207043XXXX:XXXX@sip.intervivo.net
realm = voip.project76.net
localnet = 41.0.0.0
localmask = 41.240.0.0
nat = yes

;outbound calls go here
[sip-with-london-number]
type=friend
secret=XXXX
username=0207043XXXX
host=sip.intervivo.net
insecure=very
fromuser=0207043XXXX
fromdomain=sip.intervivo.net

;soft phone client
[jamie]
type=friend
secret=XXXX
host=dynamic
nat=yes
username=jamie
disallow=all
allow=gsm
allow=ulaw
allow=alaw
context=voip





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