Hi, Edit your logger.conf, set messages in debug mode, make test incoming and outgoing calls. Copy the log in message dirz3* and post.
Goke On 3/20/10, Zeeshan Zakaria <zisha...@gmail.com> wrote: > 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 > -- Sent from my mobile device -- _____________________________________________________________________ -- 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