[asterisk-users] how to start callerid for india

cool dude cool_dudeoflko at yahoo.co.in
Sat Mar 20 12:20:05 CDT 2010


i belong to india. i am making pbx using sangoma
fxo card. i want that when ever call comes to my PSTN line i should see
the no from where call is coming. so i have to configures
chan_dahdi.conf according to my region. i checked dahdi.conf and in
that they have mentioned for india

######################################################################################################################
; Hide the name part and leave just the number part of the caller ID
; string. Only applies to PRI channels.
;hidecalleridname=yes
;
; Type of caller ID signalling in use
;     bell     = bell202 as used in US (default)
;     v23      = v23 as used in the UK
;     v23_jp   = v23 as used in Japan
;     dtmf     = DTMF as used in Denmark, Sweden and Netherlands
;     smdi     = Use SMDI for caller ID.  Requires SMDI to be enabled (usesmdi).
;
;cidsignalling=v23
;
; What signals the start of caller ID
;     ring        = a ring signals the start (default)
;     polarity    = polarity reversal signals the start
;     polarity_IN = polarity reversal signals the start, for India,
;                   for dtmf dialtone detection; using DTMF.
;                   (see doc/India-CID.txt)
;
;cidstart=polarity


so i edited chan_dahdi.conf  according to my region.

###############################################################################################################
vi chan_dahdi.conf

;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
;autogenrated on 2010-03-18
;Dahdi Channels Configurations
;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak

[trunkgroups]

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
;cidstart=ring
;cidstart=polarity
;callerid=asreceived
cidsignalling=polarity_IN
sendcalleridafter=2

;Sangoma AU100 [slot:0 bus: span:1]  <wanpipe1>
context=from-zaptel
group=0
echocancel=yes
signalling = fxs_ks
channel => 1

context=from-zaptel
group=0
echocancel=yes
signalling = fxs_ks
channel => 2

####################################################################################################

now when call comes to PSTN line i am not able to see the no. here is cli log

*CLI>     -- Starting simple switch on 'DAHDI/1-1'
[Mar 20 20:12:51] NOTICE[6659]: chan_dahdi.c:8692 ss_thread: Got event 18 (Ring Begin)...
[Mar 20 20:12:51] NOTICE[6659]: chan_dahdi.c:8692 ss_thread: Got event 17 (Polarity Reversal)...
[Mar 20 20:12:51] NOTICE[6659]: chan_dahdi.c:8692 ss_thread: Got event 17 (Polarity Reversal)...
    -- Executing [s at from-zaptel:1] Wait("DAHDI/1-1", "2") in new stack
    -- Executing [s at from-zaptel:2] GotoIfTime("DAHDI/1-1", "23:59-7:59|mon-sun|*|*?closed,s,1") in new stack
    -- Executing [s at from-zaptel:3] Dial("DAHDI/1-1", "SIP/112,60,tT") in new stack
  == Using SIP RTP CoS mark 5
    -- Called 112
    -- SIP/112-00000000 is ringing
  == Spawn extension (from-zaptel, s, 3) exited non-zero on 'DAHDI/1-1'
    -- Hungup 'DAHDI/1-1'

#################################################################################################

plz help me out.


      The INTERNET now has a personality. YOURS! See your Yahoo! Homepage. http://in.yahoo.com/
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100320/304cae2d/attachment.htm 


More information about the asterisk-users mailing list