[Asterisk-Users] SIP exten to PSTN calls

[Appan KH]

The Asrerisk config  which is tested and working is given below. The system has
1). X100P - Card
2). Two -Greadstream100 SIP Phones.

Asterisk config.

Extensions.conf
writeprotect=no

; You can include other config files, use the #include command (without the ';')
; Note that this is different from the "include" command that includes contexts within
; other contexts. The #include command works in all asterisk configuration files.
;#include "filename.conf"
; The "Globals" category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
Trunk=Zap/1
phone1=SIP/197
phone2=SIP/198
;phone3=SIP/131
everyone=${phone1}&${phone2}
[from-pstn]
exten=>_0.,1,Dial($Trunk/$everyone,13,tTmr)
exten=>_0.,2,Congestion
;exten=>_0.,1,Dial($Trunk/phone2,13,tTmr)
;exten=>_0.,2,Congestion
;ignorepat => 0




[from-sip]
;[incoming]
;exten => _0.,Dial(Zap/1/(${exten})
;exten => _0.,2,Hangup
;exten => 197,1,Dial(SIP/197,20,tr)
;exten => 197,2,Hangup
;exten => 198,1,Dial(SIP/198,20,tr)
;exten => 198,2,Hangup
;

;exten=>_.,1,Dial(Zap/1/SIP/197,20,tT)
;exten=>_.,1,Dial(Zap/1/SIP/198,20,tT)

[incoming]
;exten => 131,1,Dial(SIP/131,20,tr)
;exten => 131,2,Hangup
exten => 197,1,Dial(SIP/197,20,tr)
exten => 197,2,Hangup
exten => 198,1,Dial(SIP/198,20,tr)
exten => 198,2,Hangup


exten=>_xxxxxxxxxxx,1,Dial(${Trunk}/${EXTEN}),20,tT)
exten=>_xxxxxxxxxxx,2,Hangup

Sip.conf
;
[general]
auth=plaintext
qualify=no
nat=yes

;phone 1  Grandstream Phone
[131]
port=5060
type=peer
type=user
;context=internalsipphones
context=from-sip
host=dynamic
defaultip=212.135.237.131
canreinvite=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
;
;
;phone 2 Grandstream Phone
[198]
port=5060
type=peer
type=user
;context=internalsipphones
context=from-sip
host=dynamic
defaultip=217.37.237.198
canreinvite=yes
disllow=all
allow=ulaw
allow=alaw
allow=gsm
;
Zapata.conf[channels]
usecallerid=yes
callerid=asreceived
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
immediate=no
callprofress=no
echotraining=yes
echocancel=yes
echocancelwhenbridge=yes
switchtype=national
signalling=fxs_ks
context=from-pstn
cidsignalling=v23
cidstart=history
group=1
musiconhold=default
channel =>1
~
~
~
~
Zaptel.conf

#
loadzone = uk
#loadzone = us-old
#loadzone=gr
#loadzone=it
#loadzone=fr
#loadzone=de
#loadzone=uk
#loadzone=fi
#loadzone=jp
#loadzone=sp
#loadzone=no
defaultzone=uk
fxsks =1

The system is for UK config.

appan kh



  ----- Original Message ----- 
  From: Nil S 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Friday, September 23, 2005 9:02 AM
  Subject: Re: [Asterisk-Users] SIP exten to PSTN calls


  Hello,

  I have read your email.

  I found that you have configured X100P card and established a call from SIP exten. to SIP exten and PSTN to SIP exten.

  I have done the first part i.e. SIP exten to SIP exten and would like to do a second part. So please help me regarding this.

  I have installed Asterisk on Linux machine. So from here please guide me how i should proceed. What are the requirements? and some other details.

  Your help will be much appriciated.

  Thanks,
  Nil.

  Appan KH <appan at softswitches.net> wrote:
    Hi,
    I had configured Asterisk with the following
    1). X100P - Card
    2). Two -Greadstream100 SIP Phones.
    I am able to make calls from SIP Ext to SIP Ext and PSTN calls from outside 
    to SIP Extn.
    But I am not able to make calls from SIP Extn to PSTN out going calls-it 
    gives BT error message- The number you had dialled not recognised.
    The SIP extn is not sending the correct number.
    I will be thank full if some solutions is suggested.

    appan kh


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