Thank you Joshua, very beautiful, now webrtc is working very well.
Thanks,
Best Regards.
Ian WANG
Software Engineer
Fonality Pty Ltd(Australia)
office: +6128484 2601 ext 3007
mobile: +61402524079
On Mon, Nov 7, 2016 at 10:20 PM, Joshua Colp wrote:
> On Mon, Nov 7, 2016, at 02:14 AM, Ian Wan
On 21/10/16 10:21 AM, Mark Michelson wrote:
> I have opened https://issues.asterisk.org/jira/browse/ASTERISK-26492
> and have attached the patch there. Feel free to try it out and let me
> know on the issue how it works for you.
I've played around with your patch, and it works very well. Thanks fo
2016-11-07 16:20 GMT+01:00 Olle E. Johansson :
> Let’s extend this “how to communicate with a codec” discussion and add the
> need for feedback
> on silence. I think there’s an old mail from Matt Jordan touching this
> earlier. My code for silence suppression
> and comfort noise adds un-needed tra
2016-11-07 16:17 GMT+01:00 Joshua Colp :
> Lorenzo Miniero wrote:
>
>> Hi all,
>>
>> apologies if this has been discussed before, but I couldn't find
>> anything in the recent months on this group so I thought I'd write anyway.
>>
>> As a few others, I believe, I have been trying to find a way to
Let’s extend this “how to communicate with a codec” discussion and add the need
for feedback
on silence. I think there’s an old mail from Matt Jordan touching this earlier.
My code for silence suppression
and comfort noise adds un-needed transcoding and needed a way for a codec to
communicate st
Lorenzo Miniero wrote:
Hi all,
apologies if this has been discussed before, but I couldn't find
anything in the recent months on this group so I thought I'd write anyway.
As a few others, I believe, I have been trying to find a way to make
codec modules more aware of what's happening on the wir
2016-11-04 11:28 GMT+01:00 Lorenzo Miniero :
> Hi all,
>
> apologies if this has been discussed before, but I couldn't find anything
> in the recent months on this group so I thought I'd write anyway.
>
> As a few others, I believe, I have been trying to find a way to make codec
> modules more awa
On Mon, Nov 7, 2016 at 5:00 AM, Dan Jenkins wrote:
>
> On Fri, Nov 4, 2016 at 2:07 PM, Matt Fredrickson
> wrote:
>
>> Hey All,
>>
>> I've been thinking a lot about how working groups might work within
>> the context of the Asterisk project. Here are a few guidelines that I
>> have come up with
Hi,
while refactoring the callforwarding mechanisms for queues in our pbx i
encountered at least wrong documentation or a disappeared feature...
Until Asterisk 1.6.2 (i know, long time ago) it was possible to
distinguish between 'no agents logged in' and 'no agents available'
after executing the
On Mon, Nov 7, 2016, at 02:14 AM, Ian Wang wrote:
> hello,
> I'm trying webrtc on Asterisk 14.1.1, I've found asterisk is missing ice
> information in SDP which causing no audio issue on Chrome browser,
>
> in rtp.conf, I did configure the stun and turn server info, while in code
> res_rtp_asteris
On Fri, Nov 4, 2016 at 2:07 PM, Matt Fredrickson wrote:
> Hey All,
>
> I've been thinking a lot about how working groups might work within
> the context of the Asterisk project. Here are a few guidelines that I
> have come up with governing working groups. Some of these guidelines
> come from t
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