r);
> + // writing to socket
> + write(sockfd, cur->data.ptr,
> cur->datalen);
> }
>
>
> And we were able to see the frames on the other side of the socket.
>
> We want to ask you if there is an
if WebRTC is
desired ;)
make
```
Tested with Asterisk (comit-id 00d1c7ddd28557aa845c3522956852a60310df96).
Enjoy!
---
Dennis Guse
On Fri, Dec 23, 2016 at 12:42 PM, Dennis Guse <
dennis.g...@alumni.tu-berlin.de> wrote:
>
> The final three patches are now on Gerrit:
> * ht
. stereo) to
https://github.com/traud/asterisk-opus
Happy X-mas.
---
Dennis Guse
On Wed, Nov 30, 2016 at 10:20 PM, Richard Mudgett <rmudg...@digium.com>
wrote:
>
>
> On Wed, Nov 30, 2016 at 8:04 AM, Joshua Colp <jc...@digium.com> wrote:
>
>> On Fri, Nov 25, 2016,
...
Thus, a bridge now has two locks (ast_bridge_lock and the settings_lock),
which is some overengineering.
Is there a better solution to addressing this issue?
---
Dennis Guse
---
Dennis Guse
On Tue, Oct 25, 2016 at 5:31 PM, Joshua Colp <jc...@digium.com> wrote:
> Dennis G
ble or is this conceptually an issue?
[1]
https://github.com/asterisk/asterisk/blob/master/res/res_format_attr_opus.c#L129
[2] http://doxygen.asterisk.org/trunk/d0/d98/structast__format.html
---
Dennis Guse
On Tue, Oct 25, 2016 at 12:33 PM, Joshua Colp <jc...@digium.com> wrote:
> Dennis Guse wrot
Are we approaching the issue from a "correct" perspective?
[1] https://gerrit.asterisk.org/#/c/3524/3
[2]
https://github.com/frahaase/Asterisk_Binaural/blob/master/asterisk_modifications/include/asterisk/interleaved_stereo.h
---
Dennis Guse
On Sun, Sep 18, 2016 at 6:27 AM, Leif Madsen <
to rebuild the subsequent patches
(i.e., 3523 and 3525).
2. How can we withdraw the OPUS patch from the patch set?
https://gerrit.asterisk.org/#/c/3526/
Best,
Dennis
---
Dennis Guse
---
Dennis Guse
On Tue, Aug 16, 2016 at 11:41 AM, Dennis Guse
<dennis.g...@alumni.tu-berlin.de> wrote:
> Hel
it can be added to Asterisk
without linking it during build time?
Are there any known pitfalls or is this straight forward?
And in which folders does Asterisk search for codec modules?
Best,
Dennis
[1] https://issues.asterisk.org/jira/browse/ASTERISK-26292
---
Dennis Guse
On Fri, Aug 5, 2016 at 5
is at the moment conducted at 48kHz.
This is actually due to our use of OPUS, which always uses 48kHz for
the decoded signals.
Is this ok?
4. Is the dependency to libfftw3 an issue?
We look forward to your feedback and will spent some time preparing the patches.
---
Dennis Guse
On Wed, Jul 20, 2016
13.6.0<<):
https://github.com/steakconferencing/asterisk
In addition, we host WebRTC- based demo (the real system):
https://demo.steakconferencing.de
Best regards,
---
Dennis Guse
TU Berlin
dennis.g...@alumni.
+1
Kind regards
Dennis Guse
Quality and Usability Lab
Telekom Innovation Laboratories
TU Berlin
Ernst-Reuter-Platz 7
D-10587 Berlin, Germany
Tel: +49 30 8353 58874
Fax: +49 30 8353 58409
E-mail: dennis.g...@telekom.de
Web: www.qu.tlabs.tu-berlin.de
On Fri, Jul 18, 2014 at 8:54 PM, Stephen
-loss by adding 20ms empty frames (so
no PLC). In addition delay is quite easy to add here.
With the upcoming release of Asterisk 13, the JACK-interface is extended to
support more than 8Khz.
Best regards,
Dennis Guse
PS: To listen in such a call ChanSpy is quite useful.
[1] http://jackaudio.org
/
Testing
---
Still compiles.
Thanks,
Dennis Guse
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to the documentation to wikipedia.
Diffs
-
/trunk/main/format.c 418364
/trunk/codecs/codec_adpcm.c 418364
Diff: https://reviewboard.asterisk.org/r/3744/diff/
Testing
---
Still compiles.
Thanks,
Dennis Guse
--
_
-- Bandwidth
with jackd and puredata using G.711 and G.722 on Ubuntu 14.0.4 64bit
using Linphone and Ekiga.
Thanks,
Dennis Guse
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To UNSUBSCRIBE
.
Extension.lua is loaded as expected on startup.
Also combining extensions.conf and extensions.lua is working as expected on
startup.
Patched 11.7.0 on Ubuntu 14.04.
- Dennis Guse
On June 18, 2014, 11:27 p.m., George Joseph wrote
,
Dennis Guse
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with jackd and puredata using G.711 and G.722 on Ubuntu 14.0.4 64bit
using Linphone and Ekiga.
Thanks,
Dennis Guse
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To UNSUBSCRIBE
---
On June 14, 2014, 10:12 p.m., Dennis Guse wrote:
---
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3618
and G.722 on Ubuntu 14.0.4 64bit
using Linphone and Ekiga.
Thanks,
Dennis Guse
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_
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asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
http
415578
/trunk/apps/app_jack.c 415578
Diff: https://reviewboard.asterisk.org/r/3618/diff/
Testing
---
Checked with jackd and puredata using G.711 and G.722 on Ubuntu 14.0.4 64bit
using Linphone and Ekiga.
Thanks,
Dennis Guse
+
Diffs
-
/trunk/channels/chan_sip.c 415578
/trunk/apps/app_jack.c 415578
Diff: https://reviewboard.asterisk.org/r/3618/diff/
Testing
---
Checked with jackd and puredata using G.711 and G.722 on Ubuntu 14.0.4 64bit
using Linphone and Ekiga.
Thanks,
Dennis Guse
+
Diffs
-
/trunk/channels/chan_sip.c 415578
/trunk/apps/app_jack.c 415578
Diff: https://reviewboard.asterisk.org/r/3618/diff/
Testing
---
Checked with jackd and puredata using G.711 and G.722 on Ubuntu 14.0.4 64bit
using Linphone and Ekiga.
Thanks,
Dennis Guse
If you are in control of the SIP-Phone, you could pass additional
information via SIPAddHeader in your dialplan.
On Jun 2, 2014 10:33 AM, James Cloos cl...@jhcloos.com wrote:
Looking at app_dial.c and chan_sip.c, I get the impression that the url
in a dial string cannot get sent as part of the
AST_MODULE_LOAD_DECLINE;
}
return AST_MODULE_LOAD_SUCCESS;
}
Could it be that the pbx_lua is loaded to early and the dialplan
overwritten by some other dialplan-module that is later loaded?
PS: Work done together with Frank Haase.
Kind regards
Dennis Guse
Quality and Usability Lab
Telekom Innovation
time (and one of
my co-workers) to implement this feature.
We could start this probably in June. But as I am not an experienced
Asterisk-Developer it would be great, if you (or somebody else) could help
me in finding the best starting point...
Cheers,
Dennis
Kind regards
Dennis Guse
Quality
.
---
Dennis Guse
Kind regards
Dennis Guse
Quality and Usability Lab
Telekom Innovation Laboratories
TU Berlin
Ernst-Reuter-Platz 7
D-10587 Berlin, Germany
Tel: +49 30 8353 58874
Fax: +49 30 8353 58409
E-mail: dennis.g...@telekom.de
Web: www.qu.tlabs.tu-berlin.de
On Sun, Apr 27, 2014 at 8:50 PM
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