Hi, we are working on a new codec for asterisk. It’s early stages and
the main goal is just to be able to “hear it” using normal client end
points. So what I want is to be able to take a couple of normal G711
extensions, but have asterisk internally force a transcode to my new
codec. In this way we can easily run up a bunch of listening tests with
a wide variety of users very easily

Ok, so how do I force this transcode in the middle, ideally without
using two asterisk boxes?

I clearly need two extensions to have (say) allow=g711, which means I
need to persuade asterisk to route the call via a middle extension which
is set to only allow the new codec. What are my options to achieve
this? (meetme just occurred to me? could that work?) Any suggestions?

Thanks

Ed W

I have been breaking my head on this, can’t find a solution.

Anyone know a way to mute DTMF on SIP? I have already tried changing the
dtmfmode option and messing with different codec/dtmfmode settings but so
far, not having any luck.

Not even sure changing the codec is an option but pulling at straws at the
moment.

Thanks in advance for any assistance.

Hello,

Is anyone aware of an AMR-NB codec for Asterisk 10? The old patch floating around for older versions of Asterisk doesn’t seem to work anymore.

Best regards,
Jan Blom

Hi

Can someome give tested and proven information on Asterisk capabilities?

I have a server with 24 cores running at 2.4ghz and 16 GB RAM. How many concurrent SIP sessions I can run from single instance of Asterisk on this server? I wish to use G711 codec with echo cancel. And all calls needs to be recorded.

What will be impact on no of session when G729a is used?

Thanks & Regards,
Amit Patkar

Hi I’m developing for my company an Asterisk+t38modem+Hylafax solution.
I’m struggling in asterisk< ->t38modemv2 comunication. I’ve tried lot of
asterisk version but noone seem to function well. In asterisk 1.8.8.8
for example i can receive fax but I’m not able to send faxes due to a
“SIP/2.0 488 Not acceptable here” in asterisk 1.6 and 1.4 T38 codec
seems to be unsupported.
Asterisk 1.4.20 reports:
channel.c:3153 ast_channel_make_compatible: No path to translate from
SIP/T38modem-0-084e8dd8(256) to SIP/audiocodes_mp114-084d7858(8)
Is there an Asterisk version that could go with t38modem?
thank a lot.
I hope that someone answer becouse I’m very not so far to be unenploied…