Hi Gopal, i am using *Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V* card with tata PRI lines.
regards dhaval On Fri, Feb 4, 2011 at 3:23 PM, Gopalakrishnan A.N <[email protected]> wrote: > It seems to be you are using Sangoma T1/E1 card with echo cancellation. If > I am not wrong there is a parameter for echo cancel in the card > configuration, try disabling that because already you have enabled echo > cancel in dahdi file. > > Hope it help.:) > > On Fri, Feb 4, 2011 at 11:11 AM, DHAVAL INDRODIYA < > [email protected]> wrote: > >> Hi All, >> >> This posting regarding PRI voice optimization, on dahdi 2.1.0.4. >> >> we have more than 4 machine running on 4 port PRI card with echo >> cancellation hardware based. >> >> i have enabled echo cancel from chan_dahdi.conf using echocancel=yes, now >> more than 70% of call get good voice >> but some of calls having issue for callquality and other voice related >> issues. now my question is that is there >> any voice related parameter that we need to set for INDIA specific region >> and is ther any voice hardware tester for PRI >> that we can use and tell us our PRI [telco] provider that problem is not >> from our side. let give some idea . below are my configuration as well. >> >> >> >> # Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit >> # Zaptel Configuration File >> # >> # This file is parsed by the Zaptel Configurator, ztcfg >> # >> >> # It must be in the module loading order >> >> >> # Global data >> >> loadzone = in >> defaultzone = in >> >> >> span = 1,0,0,ccs,hdb3 >> bchan = 1-15 >> dchan = 16 >> bchan = 17-31 >> >> span = 2,0,0,ccs,hdb3 >> bchan = 32-46 >> dchan = 47 >> bchan = 48-62 >> >> span = 3,0,0,ccs,hdb3 >> bchan = 63-77 >> dchan = 78 >> bchan = 79-93 >> >> span = 4,0,0,ccs,hdb3 >> bchan = 94-108 >> dchan = 109 >> bchan = 110-124 >> >> >> >> [channels] >> language=en >> context=from-pstn >> switchtype=euroisdn >> pridialplan=local >> prilocaldialplan=local >> signalling=pri_cpe >> usecallerid=yes >> hidecallerid=no >> callwaiting=yes >> usecallingpres=yes >> callwaitingcallerid=yes >> threewaycalling=yes >> transfer=yes >> cancallforward=yes >> callreturn=yes >> relaxdtmf=yes >> echocancel=yes >> echocancelwhenbridged=yes >> echotraining=yes >> resetinterval=never >> rxgain=0.0 >> txgain=0.0 >> callgroup=1 >> pickupgroup=1 >> immediate=no >> group = 0 >> channel => 1-15 >> channel => 17-31 >> channel => 32-46 >> channel => 48-62 >> channel => 63-77 >> channel => 79-93 >> channel => 94-108 >> channel => 110-124 >> >> >> >> -- >> _____________________________________________________________________ >> -- 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 >> > > > > -- > Thank you with regards, > Gopalakrishnan A.N. > VoIP call - sip:[email protected] <sip%[email protected]> > > > > -- > _____________________________________________________________________ > -- 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
