asterisk users - Sep 2004 - How can I dial one unbusy channel of 4 available?

Hi.

I'm using asterisk as a PSTN -> SIP gateway, so that you can call to any 
of the 4 PSTN lines connected to the asterisk box from and dial your 
number, and asterisk will dial out through one of the 4 sip accounts I 
have on a SIP -> PSTN provider. I think of something like this in the 
extensions.conf

[incoming]
exten => s,1,Wait,1                     ; Wait a second, just for fun
exten => s,2,Answer                     ; Answer the line
exten => s,3,DigitTimeout,2             ; Set Digit Timeout to 5 seconds
exten => s,4,ResponseTimeout,5          ; Set Response Timeout to 10 seconds
exten => s,5,BackGround(welcome_and_dial_your_number)          ;

exten => 
_.,1,Dial(SIP/${EXTEN}@one_of_the_outgoing_sip_defined_on_sip_conf) ;*******

I dont know what to write instead of the line marked with *******. A 
multiple dial like following is not the solution I think.

exten => 
_.,1,Dial(SIP/${EXTEN}@one_outgoing_sip&SIP/${EXTEN}@other_outgoing_sip&SIP/${EXTEN}@another_outgoing_sip)


How can I know the free (or busy, is the same to me) SIP channels at any 
moment? Is there any built-in var?

Thanks in advance.

RODOLFO




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Hi Everyone,

I've been playing around with Asterisk for awhile now, and keep having 
this intermittent problem with my X100P...  Here is my setup:

Linux Kernel 2.4.26
Wildcard TDM400P (One FXS port)
Wildcard X100P (One FXO port)
Running the 1.0 release of Asterisk and the Zaptel drivers

It seems like whenever I run the server for a few hours with regular 
usage... my FXO port will get hung up on some random call.  For example, 
here is a call that has been stuck for about 20 hours:

lilith*CLI> show channels
        Channel  (Context    Extension    Pri )   State Appl.         
Data          
        Zap/1-1  (local      94252390158  2   )      Up Congestion    
(Empty)       
1 active channel(s)
lilith*CLI> show channel zap/1-1
 -- General --
           Name: Zap/1-1
           Type: Zap
       UniqueID: 1096069447.19
      Caller ID: "Main Extension" <100>
    DNID Digits: (N/A)
          State: Up (6)
          Rings: 0
   NativeFormat: 68
    WriteFormat: 4
     ReadFormat: 4
1st File Descriptor: 19
      Frames in: 3727326
     Frames out: 0
 Time to Hangup: 0
   Elapsed Time: 20h42m14s
 --   PBX   --
        Context: local
      Extension: 94252390158
       Priority: 2
     Call Group: 0
   Pickup Group: 0
    Application: Congestion
           Data: (Empty)
          Stack: 0
    Blocking in: ast_waitfor_nandfds

Whenever this happens, if I try and dial out using the extension on my 
FXS port I get these messages from the server and a rapid busy signal:

    -- Starting simple switch on 'Zap/2-1'
    -- Executing Dial("Zap/2-1", "Zap/1/2271229") in new
stack
Sep 25 13:08:18 NOTICE[573457]: app_dial.c:742 dial_exec: Unable to 
create channel of type 'Zap'
  == Everyone is busy/congested at this time
    -- Executing Congestion("Zap/2-1", "") in new stack
  == Spawn extension (local, 92271229, 2) exited non-zero on 'Zap/2-1'
    -- Hungup 'Zap/2-1'

I can easily fix the problem by restarting the Asterisk server... 
however, this is obviously less than ideal. :-(

So....  do any of you Asterisk guru's out there have a suggestion or two 
on how I can debug this problem?  I've tried looking through the system 
logs and haven't found anything particularly helpful.  Also, it doesn't 
seem to be correlated to the version of Asterisk that I'm running (I can 
reproduce it on RC1, RC2, CVS checkout, etc.)  I'm starting to wonder if 
there is a hardware problem with my X100P...  Anyway, any help or 
suggestions would be greatly appreciated!!

Thanks,
Jeremy