Mathew, Thanks Mathew. It's good to know the limitations :-) Is there any plan to add it?
On Tue, Mar 11, 2014 at 6:38 PM, Matthew Jordan <[email protected]> wrote: > On Tue, Mar 11, 2014 at 11:23 AM, Yaron Nachum <[email protected]> > wrote: > > Hi Mathew, > > The regular sip stack has 'auto' dtmfmode which behaved as I said - if > the > > remote replied with telephony event it used RFC2833 otherwise it used > > inband. > > > > Correct. There is no setting for dtmf_mode that is analogous to the > chan_sip 'auto' setting - what you configure for you endpoint today is > what it will use. > > That's not a bug, just something not existing yet. > > Matt > > -- > Matthew Jordan > Digium, Inc. | Engineering Manager > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: http://digium.com & http://asterisk.org > > -- > _____________________________________________________________________ > -- 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
