[asterisk-dev] storing INVITE fmtp and use it to send relay

[Kelvin Chua]

alex, i'll try it out, will give you feedback tomorrow.

Kelvin Chua

On Mon, Jun 29, 2015 at 9:47 PM, Kelvin Chua <kelchy at gmail.com> wrote:

> yes matt, i have it loaded
>
> Kelvin Chua
>
> On Mon, Jun 29, 2015 at 8:32 PM, Matthew Jordan <mjordan at digium.com>
> wrote:
>
>> On Mon, Jun 29, 2015 at 4:36 AM, Kelvin Chua <kelchy at gmail.com> 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
>>
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>
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