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<p>Well, I'm trying to migrate to chan_pjsip so that I don't have to
do that.<br>
</p>
<div class="moz-signature">It's so surprising that the issue so
seemingly obvious and trivial hasn't been addressed yet that I
wanted to query the collective wisdom of this list to verify my
observations. <br>
</div>
<div class="moz-signature"><br>
</div>
<div class="moz-signature">Thanks for github pointer.</div>
<div class="moz-signature"><br>
</div>
<div class="moz-signature">Michael<br>
</div>
<div class="moz-signature"><span style="font-size:10pt"><br>
</span></div>
<div class="moz-cite-prefix">On 7/5/23 16:46, <a class="moz-txt-link-abbreviated" href="mailto:asterisk@phreaknet.org">asterisk@phreaknet.org</a>
wrote:<br>
</div>
<blockquote type="cite"
cite="mid:ed311174-3c65-b48f-18f5-a2745e9853e6@phreaknet.org">On
7/5/2023 4:19 PM, Michael Ulitskiy wrote:
<br>
<blockquote type="cite">
<br>
Hi Michael,
<br>
<br>
Thanks for the reply.
<br>
<br>
I was referring to the scenario you named as 'outbound broken'.
I didn't get to look at inbound call behavior yet, as I got
stuck with inability to avoid transcoding on outbound calls.
<br>
<br>
To be more specific the scenario is as follows:
<br>
<br>
1. a phone initiates a call offering g722,g711 to asterisk<br>
2. asterisk creates outbound call to carrier offering g711 only
(carrier only supports g711)
<br>
3. carrier accepts the call and outbound call leg is now running
on g711
<br>
4. asterisk accepts a phone's call with g722 since it's allowed
on phone's endpoint and was indicated as preferred in phone's
INVITE and now initial call leg is running on g722, resulting in
transcoding
<br>
<br>
This is very disappointing. Since developers announced their
plans to drop chan_sip from future asterisk versions
<br>
<br>
</blockquote>
It's already been removed and won't be in any future major
releases.
<br>
If you still need chan_sip after removal, you can continue adding
it from out of tree and building it. I maintain a working version
of it out of tree.
<br>
<blockquote type="cite">
<br>
I was under impression that chan_pjsip has reached feature
paritiy with chan_sip.
<br>
<br>
</blockquote>
It has mostly, but not completely, no.
<br>
<blockquote type="cite">
<br>
What is needed is an ability to tell asterisk which codecs are
allowed to be included in "200 OK" asterisk sends back to the
phone. I guess we need to submit a feature request. How do we go
about it these days?
<br>
<br>
</blockquote>
I'm not sure about the particulars of this issue at all, but to
answer the question at hand, there's a repo for it:
<a class="moz-txt-link-freetext" href="https://github.com/asterisk/asterisk-feature-requests">https://github.com/asterisk/asterisk-feature-requests</a>.
<br>
</blockquote>
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