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<div class="moz-cite-prefix">perhaps a silly question ...<br>
<br>
if a channel switches from simple_bridge to native_bridge ... is
the channel switching to direct_media between the endpoints ?<br>
<br>
if so, why doesn't turn direct_media = no and
disable_direct_media_on_nat = yes switching to native_bridge off ?<br>
<br>
my pjsip.conf endpoint 7000 and 7001 <br>
<br>
[7000]<br>
type=endpoint<br>
context=outgoing<br>
disallow=all<br>
allow=alaw,ulaw,g722<br>
transport=transport-udp<br>
auth=auth7000<br>
aors=7000<br>
direct_media = no<br>
disable_direct_media_on_nat = yes<br>
<br>
[auth7000]<br>
type=auth<br>
auth_type=userpass<br>
password=xxxxxxxxxxxxxxxxxxxxxxxxx<br>
username=7000<br>
<br>
[7000]<br>
type=aor<br>
max_contacts=10<br>
qualify_frequency=60<br>
<br>
[7001]<br>
type=endpoint<br>
context=outgoing<br>
disallow=all<br>
allow=g722,alaw,ulaw<br>
transport=transport-udp<br>
auth=auth7001<br>
aors=7001<br>
direct_media = no<br>
disable_direct_media_on_nat = yes<br>
<br>
[auth7001]<br>
type=auth<br>
auth_type=userpass<br>
password=xxxxxxxxxxxxxxxxxxxxxxxxx<br>
username=7001<br>
<br>
[7001]<br>
type=aor<br>
max_contacts=10<br>
qualify_frequency=60<br>
<br>
<br>
<br>
<br>
Am 07.05.2014 07:35, schrieb Rainer Piper:<br>
</div>
<blockquote cite="mid:5369C62A.40607@soho-piper.de" type="cite">
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<div class="moz-cite-prefix">that's funny <br>
<br>
I recompiled asterisk without bridge_native_rtp.so<br>
to force asterisk to go to simple_bridge and not to
native_bridge...<br>
<br>
!!! AND THE CODEC TRANSLATION IN ASTERISK IS WORKING AGAIN !!!
juhu <br>
<br>
<br>
<br>
<br>
Am 07.05.2014 07:11, schrieb Rainer Piper:<br>
</div>
<blockquote cite="mid:5369C076.8080708@soho-piper.de" type="cite">
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<div class="moz-cite-prefix">PS.<br>
<br>
if I configure both extension 7000 and 7001 to,<br>
<br>
disallow=all<br>
allow=alaw<br>
or<br>
disallow=all<br>
allow=g722<br>
<br>
everything is fine. as long as the allowed codec is equal in
both extensions.<br>
<br>
<br>
<br>
Am 07.05.2014 07:00, schrieb Rainer Piper:<br>
</div>
<blockquote cite="mid:5369BE02.5040906@soho-piper.de"
type="cite">
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charset=ISO-8859-1">
Hi!<br>
<br>
my asterisk-12.2.0 with pjsip-2.2.0 does not translate codecs
any more. I tried every combination. silent on both sides. <br>
<br>
I compiled pjsip with no resample in pjsip.<br>
<pre>./configure --prefix=/usr --enable-shared --disable-sound <b>--disable-resample</b> --disable-video --disable-opencore-amr</pre>
is there a way to force asterisk back to do the codec
translation?<br>
<br>
Attachment:<br>
sip show channel of the calling A-Leg 7001 NativeFormats:
(g722) to the B-Leg 7000 NativeFormats: (alaw)<br>
<br>
<br>
<div class="moz-signature">-- <br>
<b>Rainer Piper</b> <br>
Integration engineer <br>
Koeslinstr. 56 <br>
53123 BONN <br>
GERMANY <br>
<br>
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<div class="moz-signature">-- <br>
<b>Rainer Piper</b> <br>
Integration engineer <br>
Koeslinstr. 56 <br>
53123 BONN <br>
GERMANY <br>
<a moz-do-not-send="true" href="http://www.soho-piper.de">www.soho-piper.de</a>
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<div class="moz-signature">-- <br>
<br>
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<br>
<br>
<div class="moz-signature">-- <br>
<b>Rainer Piper</b>
<br>
Integration engineer
<br>
Koeslinstr. 56
<br>
53123 BONN <br>
GERMANY
<br>
Phone: +49 228 97167161</div>
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