[asterisk-users] Experimenting- Sip dialing with Zap

[John Meksavan]

Asterisk Users,

  I have 3 FXO modules with the TDM400P Digium Card.  I can dial into the 
Asterisk rings my Sip phone, but dialing out with my SPA941 phone through 
the zap channel is a problem.  I keep getting this message on the Asterisk 
CLI.  What am I doing wrong? Thanks in advance.

-- Executing [103 at default:1] Dial("SIP/200-006fa300", "{Zap/g0/{EXTEN:1}") 
in new stack
[Aug 16 19:28:39] WARNING[14232]: channel.c:3209 ast_request: No channel 
type registered for '{Zap'
[Aug 16 19:28:39] WARNING[14232]: app_dial.c:1106 dial_exec_full: Unable to 
create channel of type '{Zap' (cause 66 - Channel not implemented)
  == Everyone is busy/congested at this time (1:0/0/1)
  == Auto fallthrough, channel 'SIP/200-006fa300' status is 'CHANUNAVAIL'


I used sample I found the internet just to see if this is possible.


extensions.conf

[default]
exten => 101,1,Background(tt-monkeys)
exten => 1000,1,Dial(Sip/200,20,t,r)
exten => 1000,2,Voicemail(s1000)
exten => _XXX,1,Dial({Zap/g0/{EXTEN:1})

[telewest_pstn]
exten => s,1,Dial(Sip/200,25,t,r)
exten => s,2,Voicemail
exten => s,3,Hangup


zapata.conf
[channels]
context=telewest_pstn
switchtype=national
signalling=fxs_ks
rxwink=300
channel => 2-4
adsi=yes
usecallerid=yes
cidsignalling=bell
callerid=asreceived
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=00.0
txgain=00.0
group=0
callgroup=0
pickupgroup=0
echotraining=yes
immediate=yes


zaptel.conf
loadzone=us
defaultzone=us
fxsks=2-4


Best Regards,
John

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