[asterisk-users] FXS ports on TDM410P card...

[Michael C. Robinson]

[Oct 15 01:48:02] NOTICE[3747] channel.c: Dropping incompatible voice
frame on SIP/2006-00000000 of format ulaw since our native format has
changed to 0x8 (alaw)
[Oct 15 01:48:49] WARNING[3750] pbx.c: Channel 'DAHDI/1-1' sent into
invalid extension 's' in context 'default', but no invalid handler
----------------------------------------------------------------------
I have the code to set up an extension for toggling Telco pass through
working I think.  What isn't working is the pass through.  I get the
above error messages when I try to call the POTS line connected to
DAHDI/1 from my Comcast line.  

I'm noticing other warning messages cropping up about this file or that
file not existing and modules not loading, but mostly the system seems
to be working so I'm wondering if these warnings are relevant.  I'm
using Asterisk 1.8.

I think that [from-pstn] isn't working...

For those who don't know what I'm after, I'm trying when a phone company
call comes in to ring SIP phones and local FXS lines on my TDM410P.  The
purpose of the toggle is to be able to disable this feature.  Sometimes,
I really want to use this system as a private intercom system where at
other times, ringing remote SIP phones for an incoming telephone company
call might be needed.  Say you are at extension 2000 or 2002, SIP phones
in other buildings, and you want or need to be able to receive calls
from the PSTN.

I'm in the U.S., under the [external] section am I blocking long
distance outgoing phone calls?  In the U.S., you dial 1 and then
the number for long distance.  Essentially, what I need to do is 
block dialing 1 and then a number with the exception of 1-800 or 
1-866.

Thank you for taking the time to look at my questions and
information ;-)

My current extensions.conf file in it's entirety follows:
---------------------------------------------------------------------
[globals]
CENTURYLINK=DAHDI/1
COMCAST=DAHDI/2
ANDREWROOM=DAHDI/3
SERVERROOM=DAHDI/4
WIDE_PBX=SIP/2000&SIP/2002&SIP/2006&SIP/2007&SIP/2008&SIP/2009&
${SERVERROOM}&${ANDREWROOM}
INSIDE_PBX=SIP/2006&SIP/2007&SIP/2008&SIP/2009&${SERVERROOM}&
${ANDREWROOM}
OUTSIDE_PBX=SIP/2000&SIP/2002
TELCO_ON=0
PSTN_THROUGH=${SERVERROOM}&${ANDREWROOM}&SIP/2000&SIP/2002



[external]
exten => _9NXXNXXXXXX,1,Dial(${CENTURYLINK}/${EXTEN:1})
exten => _8NXXNXXXXXX,1,Dial(${COMCAST}/${EXTEN:1})



[my-phones]
exten => i,1,Playback(/var/lib/asterisk/sounds/custom/extns-list)
exten => i,n,Hangup()

exten => 2000,1,Dial(SIP/2000,40)
 same => n,VoiceMail(2000,u)

exten => 2002,1,Dial(SIP/2002,40)
 same => n,VoiceMail(2002,u)

exten => 2004,1,Dial(SIP/2004,40)
 same => n,VoiceMail(2004,u)

exten => 2006,1,Dial(SIP/2006,40)
 same => n,VoiceMail(2006,u)

exten => 2007,1,Dial(SIP/2007,40)
 same => n,VoiceMail(2007,u)

exten => 2008,1,Dial(SIP/2008,40)
 same => n,VoiceMail(2008,u)

exten => 2009,1,Dial(SIP/2009,40)
 same => n,VoiceMail(2009,u)

exten => 2010,1,Dial(${SERVERROOM},40)
 same => n,VoiceMail(2009,u)

exten => 2010,1,Dial(${SERVERROOM},40)
 same => n,VoiceMail(2010,u)

exten => 2011,1,Dial(${ANDREWROOM},40)
 same => n,VoiceMail(2011,u)

exten => 2012,1,Dial(${WIDE_PBX},40)

exten => 2013,1,Dial(${INSIDE_PBX},40)

exten => 2014,1,Dial(${OUTSIDE_PBX},40)

exten => 2015,1,GoToIf($[${TELCO_ON}=1]?2:5)
; 2 turns off "telco_on"
exten => 2015,2,Set(GLOBAL(TELCO_ON)=0)
exten => 2015,3,Playback(/var/lib/asterisk/sounds/custom/telco-off)
exten => 2015,4,hangup()
; 5 turns on "telco_on"
exten => 2015,5,Set(GLOBAL(TELCO_ON)=1)
exten => 2015,6,Playback(/var/lib/asterisk/sounds/custom/telco-on)
exten => 2015,7,hangup()

exten => 2999,1,VoiceMailMain(${CALLERID(num)},s)

[from-pstn]
exten => s,1,GoToIf($[${TELCO_ON}=1]?2:3)
; 2 rings all phones
exten => s,2,Dial(${PSTN_THROUGH},40)
exten => s,3,Hangup()

include => external
include => from-pstn