asterisk users - Jul 2004 - TDM400P Internal Extenion Config

Hopefully someone here can save my sanity. I have been trying to solve 
this problem for days now, but just cant put my finger on it. Im new to 
* so I have probably done something stupid!

I have a TDM400P with one FXO module and a FXS module. The main problem 
I have is not being able to get the extension attached to the FXS module 
to ring or be able to make calls. It gets a dialtone fine but I guess 
this doesnt really mean all that much.

If I dial the extension I just get a 404 error on the phone 
(Grandstream), but no errors at all on the console. I am using 
CVS-HEAD-07/14/04.  Here is a snippet of what I have in the various 
config files.


zaptel.conf

loadzone=au
defaultzone=au
fxsks=1
fxoks=2

zapata.conf

[channels]
context=incoming
switchtype=national
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
txgain=0.0
rxgain=0.0
group=1
callgroup=1
pickupgroup=1-4
immediate=no
busydetect=yes
busycount=7

callerid=asreceived
channel => 1
context=internal
group=2

signalling=fxo_ks
callerid="Fax" <310>

channel => 2


extensions.conf

[general]
static=yes
writeprotect=no

[globals]

[sip]
exten => 301,1,Dial(SIP/Nick,20,tr)
exten => 302,1,Dial(SIP/Sharon,20,tr)
exten => 1000,1,Dial(SIP/Nick&SIP/Sharon,20,tr)
exten => 302,2,VoiceMail,u302
exten => 301,2,VoiceMail,u301
exten => 1000,2,VoiceMail,u9999
exten => 1000,102,VoiceMail,b9999
exten => 1001,1,Ringing
exten => 1001,2,Wait(2)
exten => 1001,3,VoicemailMain
include => outgoing

[incoming]
exten => s,1,Dial(SIP/Nick&SIP/Sharon,20,tr)

[outgoing]
exten => _7.,1,Dial(IAX2/login:passwd@XXX.XXX.XXX>XXX/${EXTEN:1})
exten => 5.,1,Dial,Zap/1/${EXTEN:1}

[9103]
exten => 21060,1,Dial(SIP/Nick)
exten => 21062,1,Dial(SIP/Sharon)

[internal]
exten => 310,1,Dial,Zap/2



If I try to make any calls from the extension connected to the fxs 
module i just get what sounds like a busy tone. Looking at the console 
it generally give the error "zt_set_hook: zt hook failed Device or 
resource busy". It only gives this error when it goes off hook and 
number dialed.

Only other information I can provide is a couple errors when asterisk 
start up. I get the following:

"Unable to open /dev/dsp: No such device"
"Unable to get our IP address, Skinny disable"
"Ignoring switchtype"

Have not been able to dig out vast amounts of information on the above, 
but what I have found didnt seem to point to my problem, but then what 
do I know!

If anyone can help I would appreciate it! I'm going crazy here!

Kind regards
Nick

Forgot to mention, both modules are show in ztcfg fine, see below:

Zaptel Configuration
=====================

Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)

2 channels configured.

and zap show channel 2 give the following:

Channel: 2
File Descriptor: 19
Span: 1
Extension:
Dialing: no
Context: internal
Caller ID string: "Fax" <310>
Destroy: 0
Signalling Type: FXO Kewlstart
Owner: <None>
Real: <None>
Callwait: <None>
Threeway: <None>
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps, currently OFF
Actual Confinfo: Num/0, Mode/0x0000
Actual Confmute: No
Actual Hookstate: Onhook


Okay, so now I'm going to lie down in a dark room.

Cheers

Nick Cobley wrote:
> Hopefully someone here can save my sanity. I have been trying to solve 
> this problem for days now, but just cant put my finger on it. Im new 
> to * so I have probably done something stupid!
>
> I have a TDM400P with one FXO module and a FXS module. The main 
> problem I have is not being able to get the extension attached to the 
> FXS module to ring or be able to make calls. It gets a dialtone fine 
> but I guess this doesnt really mean all that much.
>
> If I dial the extension I just get a 404 error on the phone 
> (Grandstream), but no errors at all on the console. I am using 
> CVS-HEAD-07/14/04.  Here is a snippet of what I have in the various 
> config files.
>
>
> zaptel.conf
>
> loadzone=au
> defaultzone=au
> fxsks=1
> fxoks=2
>
> zapata.conf
>
> [channels]
> context=incoming
> switchtype=national
> signalling=fxs_ks
> usecallerid=yes
> hidecallerid=no
> callwaiting=no
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> txgain=0.0
> rxgain=0.0
> group=1
> callgroup=1
> pickupgroup=1-4
> immediate=no
> busydetect=yes
> busycount=7
>
> callerid=asreceived
> channel => 1
> context=internal
> group=2
>
> signalling=fxo_ks
> callerid="Fax" <310>
>
> channel => 2
>
>
> extensions.conf
>
> [general]
> static=yes
> writeprotect=no
>
> [globals]
>
> [sip]
> exten => 301,1,Dial(SIP/Nick,20,tr)
> exten => 302,1,Dial(SIP/Sharon,20,tr)
> exten => 1000,1,Dial(SIP/Nick&SIP/Sharon,20,tr)
> exten => 302,2,VoiceMail,u302
> exten => 301,2,VoiceMail,u301
> exten => 1000,2,VoiceMail,u9999
> exten => 1000,102,VoiceMail,b9999
> exten => 1001,1,Ringing
> exten => 1001,2,Wait(2)
> exten => 1001,3,VoicemailMain
> include => outgoing
>
> [incoming]
> exten => s,1,Dial(SIP/Nick&SIP/Sharon,20,tr)
>
> [outgoing]
> exten => _7.,1,Dial(IAX2/login:passwd@XXX.XXX.XXX>XXX/${EXTEN:1})
> exten => 5.,1,Dial,Zap/1/${EXTEN:1}
>
> [9103]
> exten => 21060,1,Dial(SIP/Nick)
> exten => 21062,1,Dial(SIP/Sharon)
>
> [internal]
> exten => 310,1,Dial,Zap/2
>
>
>
> If I try to make any calls from the extension connected to the fxs 
> module i just get what sounds like a busy tone. Looking at the console 
> it generally give the error "zt_set_hook: zt hook failed Device or 
> resource busy". It only gives this error when it goes off hook and 
> number dialed.
>
> Only other information I can provide is a couple errors when asterisk 
> start up. I get the following:
>
> "Unable to open /dev/dsp: No such device"
> "Unable to get our IP address, Skinny disable"
> "Ignoring switchtype"
>
> Have not been able to dig out vast amounts of information on the 
> above, but what I have found didnt seem to point to my problem, but 
> then what do I know!
>
> If anyone can help I would appreciate it! I'm going crazy here!
>
> Kind regards
> Nick
>
>

On Mon, 2004-07-19 at 07:13, Nick Cobley wrote:
> If I dial the extension I just get a 404 error on the phone 
> (Grandstream), but no errors at all on the console. I am using 
> CVS-HEAD-07/14/04.  Here is a snippet of what I have in the various 
> config files.
Welcome to SIP. Dialtone is local to your phone and is not dependent on
proper config. Hope that helps put you on the correct step to fix that
problem.
-- 
Steven Critchfield  <critch@basesys.com>

Thanks Steve,

The SIP handsets are working find as I can make calls to other handsets 
as well as receive incoming calls via the FXO module. So all is good there.

Cheers
Nick

Steven Critchfield wrote:
>On Mon, 2004-07-19 at 07:13, Nick Cobley wrote:
>
>  
>
>>If I dial the extension I just get a 404 error on the phone 
>>(Grandstream), but no errors at all on the console. I am using 
>>CVS-HEAD-07/14/04.  Here is a snippet of what I have in the various 
>>config files.
>>    
>>
>
>Welcome to SIP. Dialtone is local to your phone and is not dependent on
>proper config. Hope that helps put you on the correct step to fix that
>problem.
>  
>

On Mon, 19 Jul 2004 20:13:09 +0800, Nick Cobley <info@nvworld.net>
wrote:
> Hopefully someone here can save my sanity. I have been trying to solve
> this problem for days now, but just cant put my finger on it. Im new to
> * so I have probably done something stupid!
Only a config issue I'm sure
 
> [sip]
> exten => 301,1,Dial(SIP/Nick,20,tr)
> exten => 302,1,Dial(SIP/Sharon,20,tr)
> exten => 1000,1,Dial(SIP/Nick&SIP/Sharon,20,tr)
> exten => 302,2,VoiceMail,u302
> exten => 301,2,VoiceMail,u301
> exten => 1000,2,VoiceMail,u9999
> exten => 1000,102,VoiceMail,b9999
> exten => 1001,1,Ringing
> exten => 1001,2,Wait(2)
> exten => 1001,3,VoicemailMain
> include => outgoing
add here 
include => internal  ; allow sip to dial 310
> [incoming]
> exten => s,1,Dial(SIP/Nick&SIP/Sharon,20,tr)
> 
> [outgoing]
> exten => _7.,1,Dial(IAX2/login:passwd@XXX.XXX.XXX>XXX/${EXTEN:1})
> exten => 5.,1,Dial,Zap/1/${EXTEN:1}
> 
> [9103]
> exten => 21060,1,Dial(SIP/Nick)
> exten => 21062,1,Dial(SIP/Sharon)
> 
> [internal]
> exten => 310,1,Dial,Zap/2
include => sip ; allow internal to dial sip phone
> 
Try those changes and see how you get on


Jason