[asterisk-users] PRI voice optimization

[DHAVAL INDRODIYA]

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 <saigop at gmail.com> 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 <
> dhaval.it01034 at gmail.com> 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
>>
>>
>>
>> --
>> _____________________________________________________________________
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>
>
>
> --
> Thank you  with regards,
> Gopalakrishnan A.N.
> VoIP call - sip:saigop at gtalk2voip.com <sip%3Asaigop at gtalk2voip.com>
>
>
>
> --
> _____________________________________________________________________
> -- 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
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
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