Hi, Thanks for this information, but it is not working for both the issues, I have tried with the configuration with cidsignalling, cidstart etc.. Can any one provide more help for this.
Thanks, Max Alex Voip Developer On Mon, Aug 9, 2010 at 5:31 PM, asteriskguru asteriskguru < beaasteriskg...@gmail.com> wrote: > Hi max, > Have look on my blog regarding this. > > > http://ashikalim.blogspot.com/2010/08/config-asterisk-india-pstn-lines_09.html > > Thanks, > Ashik > > On Sat, Aug 7, 2010 at 11:15 AM, Max Alex <max.aster...@gmail.com> wrote: > >> Hi All, >> I have Sangoma A200 Card installed on my system, >> I have centos 5.5 with 64 bit, >> Here are the description for asterisk and dahdi. >> Asterisk 1.6..2.9 >> Dahdi: 2.3.0.1 >> I have two issues with dahdi >> 1) I am not getting full callerid on my phones from sangoma card to >> asterisk users. if i am connecting analog phone directly then i am getting >> callerid properly. >> I am in india and using Airtel Connection, I have set variables in >> chan_dahdi.conf as well for callerid but the not getting full digits in >> callerid, >> it is coming with 8 digits only. >> 2) Another issue is when I am hanging up the phone from inbound or >> outbound from the dahdi channel, it takes 5-6 seconds to dropping the call. >> >> Here are the confguration file for chan_dahdi.conf >> ------------------------------------- >> ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit >> ;autogenrated on 2010-07-30 >> ;Dahdi Channels Configurations >> ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak >> >> [trunkgroups] >> >> [channels] >> context=default >> usecallerid=yes >> callerid=asreceived >> hanguponpolarityswitch=yes >> answeronpolarityswitch=yes >> ;cidstart=ring >> cidstart=polarity_IN >> ;cidsignalling=dtmf >> cidsignalling=dtmf >> 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 >> useincomingcalleridondahditransfer=yes >> ;callerid=asreceived >> >> ;Sangoma AFT-A200 [slot:4 bus:2 span:1] <wanpipe1> >> context=from-internal >> group=1 >> echocancel=yes >> callerid=asreceived >> signalling = fxo_ks >> channel => 1 >> >> context=from-internal >> group=1 >> echocancel=yes >> callerid=asreceived >> signalling = fxo_ks >> channel => 2 >> >> context=from-zaptel >> group=0 >> echocancel=yes >> callerid=asreceived >> signalling = fxs_ks >> channel => 3 >> >> context=from-zaptel >> group=0 >> echocancel=yes >> callerid=asreceived >> signalling = fxs_ks >> channel => 4 >> ------------------------------- >> Please hemp me for this issues. >> >> Thanks, >> Max Alex >> Voip Developer >> >> >> -- >> _____________________________________________________________________ >> -- 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 >
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