Hi, thanks for your replay Alex:
Right now a have an Asterisk server on a Dell Optiplex GX110 (PIII 666MHz, 320 RAM) with no soundcard.
With an X100P clone card (an ambient modem).
Everything looks good, I've been able to make local calls trough PSTN, IAX, SIP.
I only have 1 POTS line, and 4 SIP softphones (X-lite)running all right.
The only problem so far I have noticed (or realized of :P), it is that i can make calls
to cellularphone numbers, * tries to connect but i get redirected to the emergency service number 066.
I don't think it is because of my dialplan, eventhough I tried several configurations. Anyways here is part of the dialplan
where my softphones make calls:
;;# Llamadas salientes [outgoing] #;
[outgoing]include = toPSTNinclude = iaxtelinclude = fwd-iax
; - PSTN
[toPSTN] ; Permite hacer llamadas locales (7-digitos sin contar 9)ignorepat = 9
exten = _92XX,1,NoOp(Call for ${EXTEN:1})exten = _92XX,2,Dial(Zap/1/${EXTEN:1})
exten = _904466,1,NoOp(Call for (${EXTEN:1}) ;Llamadas a Celularexten = _904466,2,Dial(Zap/1/ww${EXTEN:1})
; - IAXTEL
[iaxtel]exten = _1700XXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]
)exten = _1888NXX,1,Dial( IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]
)exten = _1877NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]
)exten = _1866NXX,1,Dial( IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]
)exten = _1800NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]
)
; - FWD
[fwd-iax]exten = _3.,1,SetCallerId,${FWDCIDNAME}exten = _3.,2,Dial(
IAX2/${FWDNUMBER}:[EMAIL PROTECTED]/${EXTEN:1},60,tr) exten = _3.,3,Congestion
;#;# Softphone x-lite #;#
[x-lite] ; Note: SIP extensions are defined here as 66 followed by any two digitsinclude = defaultinclude = serviciosinclude = outgoing
exten = 6600,1,NoOp(Llamada saliente maneja IAX2)exten = 6600,2,Macro(dial,kano00,IAX2/kano00,20,tr)
exten = _X,1,NoOp(Llamada saliente maneja SIP)exten = _X,2,Macro(dial,667${EXTEN},SIP/667${EXTEN},20,tr)
Allsoftphones working are SIP, and are directed to the [x-lite] context.
This is my zapata.conf:
[channels]language=escontext=incomingsignalling=fxs_ks.usecallerid=yeshidecallerid=nocallwaiting=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=yescallreturn=yes
echocancel=yesechocancelwhenbridged=yesechotraining=800rxgain=0.0txgain=25.0
group=1pickupgroup=1immediate=yesmusiconhold=defaultrelaxdtmf=yes ; Relajar el DTMF, poner si asterisk salta o duplica algún DTMF, ; dando lugar a un número incorrecto.channel = 1
And my simple configuration file,
zaptel.conf:
loadzone=mxdefaultzone=mxfxsks=1
As you can see this aren't complicated configurations because i only have 1 X100P card, and I am currently using little extensions.Also, I am not using AMP but I'm thinking to installing it over my current installation.I installed asterisk andzaptel from instructions i got from several documentations sites (voip-info wiki, digium, etc).
Well, I hope this info can help to look down the problem. Thanks again,
Regards,
Claudio
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
