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

[Kelvin Chua]

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 <kelchy at gmail.com> 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 <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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