[EMAIL PROTECTED] wrote:
On 30 Aug 2004 at 0:26, Steve Underwood wrote:
[EMAIL PROTECTED] wrote:
Why doesn't asterisk clock to the 1000 interrupts per second instead of the incoming audio? Were there no interrupts available when it started? Even if you had no card you could use the ztdummy module and even though that might be off by a bit, surely it'd sound better than a connection which is experiencing packet loss?
How much work would be required to change this? I guess it couldn't really be an option because of the totally different structure...
Would it be possible for one person to make those changes or would it require the authors of all modules to recode?
I haven't even completed by soft fax machine, and you are trying to be it completely useless. :-) Think about that. What you are suggesting is not really a satisfactory solution to anything, but certainly breaks things. :-\
Is this English?!
my soft fax?
make it completely useless?
Okay, I think I understand you now...
This surely wouldn't concern your code unless your code does it's transmission via IAX, SIP, OpenH.323 etc?
And unless I'm gravely mistaken fax won't work over IP anyway...
Is this a well thought out response?
FAX won't work over IP?
Doesn't changing the timing in the core of * affect the PSTN channels as well as the IP ones?
Doesn't everything - caller ID, my soft fax machine, SMS, etc. - that works within * all go through the * core?
Won't this screw up everything just to keep you happy?
Won't this actually fail to keep you happy, since you don't seem to have thought through the whole jitter buffering issue, anyway?
So many questions. So few meaningful answers :-)
Regards, Steve
_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
