that's funny
I recompiled asterisk without bridge_native_rtp.so
to force asterisk to go to simple_bridge and not to native_bridge...
!!! AND THE CODEC TRANSLATION IN ASTERISK IS WORKING AGAIN !!! juhu
Am 07.05.2014 07:11, schrieb Rainer Piper:
PS.
if I configure both extension 7000 and 7001 to,
disallow=all
allow=alaw
or
disallow=all
allow=g722
everything is fine. as long as the allowed codec is equal in both
extensions.
Am 07.05.2014 07:00, schrieb Rainer Piper:
Hi!
my asterisk-12.2.0 with pjsip-2.2.0 does not translate codecs any
more. I tried every combination. silent on both sides.
I compiled pjsip with no resample in pjsip.
./configure --prefix=/usr --enable-shared --disable-sound*--disable-resample*
--disable-video --disable-opencore-amr
is there a way to force asterisk back to do the codec translation?
Attachment:
sip show channel of the calling A-Leg 7001 NativeFormats: (g722) to
the B-Leg 7000 NativeFormats: (alaw)
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
------------------------------------------------------------------------
------------------------------------------------------------------------
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
www.soho-piper.de <http://www.soho-piper.de>
------------------------------------------------------------------------
------------------------------------------------------------------------
--
*Rainer Piper*
Integration engineer
Koeslinstr. 56
53123 BONN
GERMANY
www.soho-piper.de <http://www.soho-piper.de>
------------------------------------------------------------------------
NOC +49 228 97167161 - sip.soho-piper.de
NOC +882 990111550 via e164.org International Network
------------------------------------------------------------------------
NOC +49 2247 9064188 - sip.tefonix.de - D293
NOC +882 990045450 via e164.org International Network
------------------------------------------------------------------------
--
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