Works alex, i added a couple more parameters to copy from the original Invite fmtp. hope this find its way into trunk soon.
Kelvin Chua On Mon, Jun 29, 2015 at 9:58 PM, Kelvin Chua <[email protected]> wrote: > alex, i'll try it out, will give you feedback tomorrow. > > Kelvin Chua > > On Mon, Jun 29, 2015 at 9:47 PM, Kelvin Chua <[email protected]> wrote: > >> yes matt, i have it loaded >> >> Kelvin Chua >> >> On Mon, Jun 29, 2015 at 8:32 PM, Matthew Jordan <[email protected]> >> wrote: >> >>> On Mon, Jun 29, 2015 at 4:36 AM, Kelvin Chua <[email protected]> wrote: >>> > Guys, >>> > >>> > just tried asterisk13 and added seanbrights' patch for opus. >>> > >>> > incoming INVITE has fmtp ------> >>> > maxplaybackrate=8000;sprop-maxcapturerate=8000 >>> > but INVITE to my registered peer is ----------> >>> > maxplaybackrate=48000;sprop-maxcapturerate=48000 >>> > >>> > it should not even have to load up the opus patch because it is just a >>> > passthrough >>> > have you changed anything to chan_sip.c to make this work? >>> > >>> >>> Do you have res_format_attr_opus loaded? >>> >>> -- >>> Matthew Jordan >>> Digium, Inc. | Director of Technology >>> 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 -- >>> >>> asterisk-dev mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-dev >>> >> >> >
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