Hi,
my scenario is below
analog phone (10 to 99)------> pbx------>(77)asterisk--------> jitsi(2000)
i have analog telephone interface numbered 77 attached with asterisk and
other sip user is 2000 on jitsi.
I can call from any number from 10 to 99(in intercom) on 77 and ivr
response will come then i can typed 2000# and call go to 2000 named user
in asterisk.
Now my problem is when i am calling from 10 to 99 (any number) this number
should display to sip 2000's user. But its not showing to user. Its shows
asterisk@my_asterisk_server_ip.
my config. as follow
extension.conf
exten => s,1,Goto(phrase-menu,s,1)
[phrase-menu]
exten => s,1,Answer()
exten => s,2,Wait(1)
exten => s,3,Read(PHRASEID,/var/lib/asterisk/sounds/custom/soip)
exten => s,4,Wait(2)
exten => s,5,Set(CALLERID(num,CID)=${CALLERID})
exten => s,6,Dial(SIP/${PHRASEID},40,tT)
exten => h,1,Hangup()
and in chan_dahdi.conf
; General options
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
cidsignalling=dtmf
cidstart=polarity
callerid=asreceived
rxgain=0.0
txgain=0.0
;FXO Modules
group=1
echocancel=yes
signalling=fxs_ks
context=default
channel=1-20
#include dahdi-channels.conf
any help
thanks..
Do not bother about below message. That is auto-generated by my mail
server.
--
With Warm Regards
Harish Mandowara
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