[asterisk-users] ISP< ->Asterisk <-> ATA <->DIALUP

Don Fanning don at 00100100.net
Mon Jun 29 21:46:33 CDT 2009


Not true...

You can provided you disable data compression (AT&K0) on your modem.   
Reason?  Because a codec is already compressed.  Adding compression at
the modem level to an already compressed bitstream == lost bits.  I call
all over the world all the time using asterisk/sip/ulaw with decent bit
rates.



Alex Balashov wrote:
> Without getting into a lot of detail, this will not work.  Period.
>  You just can't do reliable modem passthrough with VoIP in most cases,
> some clever proprietary hacks notwithstanding.
>
> To the extent it is possible, nobody is going to "send you the
> procedure.". This list is for specific answers to specific questions.
>
> --
> Sent from mobile device
>
> On Jun 29, 2009, at 10:47 AM, Vidura Senadeera <vidurased at gmail.com
> <mailto:vidurased at gmail.com>> wrote:
>
>> Hellow, 
>>
>> / I have a problem with dial up signalling. currently I have
>> configured asterisk server and E1 card to ISP. then other side I am
>> having ATA to PC for connecting internet through DialUP connection.
>> is it possible and please send me the procedure how I can do it ??  /
>>
>> ISP< <-> Asterisk <-> ATA <-> DIALUP
>> -- 
>> Thanks & Regards,
>> Vidura Senadeera,
>> Sri Lanka.
>> msn/yahoo/skype Ids - vidurased
>> _______________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> 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 --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users



More information about the asterisk-users mailing list