[asterisk-dev] fallback to audio faxing when T.38 INVITE fail with 488/606 Not acceptable

Kevin P. Fleming kpfleming at digium.com
Tue Feb 9 09:20:45 CST 2010


Kristijan Vrban wrote:

> Again http://tools.ietf.org/html/draft-ietf-sipping-realtimefax-01#section-6.2
> 
> "The terminating gateway SHOULD react by proposing a fallback to G.711
> fax pass-through with special codec characteristics - -silence suppression OFF"

Keep in mind that you are posting references to a draft that expired
seven years ago; it is not realistic to expect any SIP agent to be
'conforming' to that draft since it never progressed towards publication
as any sort of RFC (although we're working in the SIP Forum on making
something like that finally happen).

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpfleming at digium.com
Check us out at www.digium.com & www.asterisk.org



More information about the asterisk-dev mailing list