Does your regular phone shows callerid on this line. If the service provider is sending the callerid, asterisk doesn't have to do anything special to retrieve it.
-- Zeeshan On 2010-03-20 1:25 PM, "cool dude" <cool_dudeof...@yahoo.co.in> wrote: 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 [...@from-zaptel:1] Wait("DAHDI/1-1", "2") in new stack -- Executing [...@from-zaptel:2] GotoIfTime("DAHDI/1-1", "23:59-7:59|mon-sun|*|*?closed,s,1") in new stack -- Executing [...@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. ------------------------------ Your Mail works best with the New Yahoo Optimized IE8. Get it NOW!<http://in.rd.yahoo.com/tagline_ie8_new/*http://downloads.yahoo.com/in/internetexplorer/> . -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users