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<p>The Real-Time Text feature of Asterisk does not work with PJSIP.
Or at least it is not documented how its redundant transport
support is configured.<br>
See:
<a class="moz-txt-link-freetext" href="https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=4260034">https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=4260034</a>
for how it once worked.</p>
<p>(There are bugs in the release 11 and 13 implementations of
redundant transmission of real-time text with chan_sip as well,
but it worked in earlier releases. )<br>
</p>
<p><br>
</p>
<br>
<div class="moz-cite-prefix">Den 2017-10-08 kl. 16:55, skrev James
Finstrom:<br>
</div>
<blockquote type="cite"
cite="mid:CAE3BMY2g3d7AZWPKGshQR-xSPvMWqG0rt2xyT0OmB=M_O-ac6w@mail.gmail.com">
<div dir="auto">One does not simply depricate a sip stack.
<div dir="auto"><br>
</div>
<div dir="auto">Ok so at devcon there was a discussion of
depricating chan_sip. This may sound a lot worse than it
actually is. Chan_sip has been essentially untouched in 4ish
years. It does not receive bug fixes. It is just sort of a
barge floating in the ocean. </div>
<div dir="auto"><br>
</div>
<div dir="auto">So one of the things that is needed to finally
put Chan sip to bed is feature parody. Someone brought up
CCSS.</div>
<div dir="auto"><br>
</div>
<div dir="auto">What features do you feel you would lose going
from chan_sip to pjsip.</div>
<div dir="auto"><br>
</div>
<div dir="auto">Are there any bugs in pjsip that keep you from
migrating? </div>
</div>
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</blockquote>
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