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<div class="moz-cite-prefix">I have done exactly the same way from
chan_sip to res_pjsip. It was a great stress!<br>
<br>
I talked a lot with Joshua Colp. Big thanks to him for the help. <br>
But even now in the res_pjsip much of the functionality is
unavailable. I wrote a mail to the mailing lists, created issues.<br>
Much of the requested is not implemented and it is not changing.<br>
<br>
Moreover, I couldn't go with 13.9 for newer versions. The reason
for this is unexplained fall immediately after loading in several
different places. <br>
Perhaps it's the personal features of my "old" operating system
(slackware). <br>
<br>
I will be very happy if there is such a purpose, and a working
group that will implement it.<br>
<br>
05.10.2016 13:14, Ross Beer пишет:<br>
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<p>Hi</p>
<p><br>
</p>
<p>I have spent over a year migrating from chan_sip (1.8) to
chan_pjsip (13) and it has been stressful. </p>
<p><br>
</p>
<p>However, there is light at the end of the tunnel. When first
migrating Asterisk would crash around 20 times a day or more.
However, by investing time and money into resolving the
segfaults, database issues and task managers I feel that the
new stack is stable with the odd bug still remaining. The most
common crash I get from the stack at the moment is due to TLS
connections, which the PJSIP team are currently working on and
I am assured there will be a patch in the coming days.</p>
<p><br>
</p>
<p>From experience, I can say that chan_pjsip is more scalable
and efficiently uses server resources compared to chan_sip. It
is the way forward!</p>
<p><br>
</p>
<p>I would welcome a working group to manage the migration from
chan_sip to chan_pjsip as there are still features in chan_sip
that have not been implemented in chan_pjsip. I would also
welcome additional features such as '<span>Device Feature Key
Synchronization' (as-feature-event).</span></p>
<p><span><br>
</span></p>
<p><span>At present, there are a few undocumented features, such
as the sorcery configuration:</span></p>
<p><span><br>
</span></p>
<blockquote style="margin: 0 0 0 40px; border: none; padding:
0px;">
<p><span><span>endpoint=realtime,ps_endpoints,<b>allow_unqualified_fetch=error</b></span></span></p>
<p><span><span><b><br>
</b></span></span></p>
</blockquote>
The above stops a full database query that loads every single
endpoint at startup, which can cause overload on systems with a
number of endpoints. Therefore documentation covering the whole
sip stack and features would help people migrate easier.<br>
<p><br>
</p>
<p>Finally, I would like to thank everyone who has been working
on ironing out the chan_pjsip bugs.</p>
<p><br>
</p>
<p>Ross</p>
<br>
<br>
<div style="color: rgb(0, 0, 0);">
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<hr tabindex="-1" style="display:inline-block; width:98%">
<div id="x_divRplyFwdMsg" dir="ltr"><font
style="font-size:11pt" face="Calibri, sans-serif"
color="#000000"><b>From:</b>
<a class="moz-txt-link-abbreviated" href="mailto:asterisk-dev-bounces@lists.digium.com">asterisk-dev-bounces@lists.digium.com</a>
<a class="moz-txt-link-rfc2396E" href="mailto:asterisk-dev-bounces@lists.digium.com"><asterisk-dev-bounces@lists.digium.com></a> on behalf
of Olle E. Johansson <a class="moz-txt-link-rfc2396E" href="mailto:oej@edvina.net"><oej@edvina.net></a><br>
<b>Sent:</b> 05 October 2016 10:42<br>
<b>To:</b> Asterisk Developers Mailing List<br>
<b>Cc:</b> Olle E Johansson<br>
<b>Subject:</b> Re: [asterisk-dev] Viva Chan_Sip, may it
rest in peace</font>
<div> </div>
</div>
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<font size="2"><span style="font-size:10pt;">
<div class="PlainText">Hi!<br>
<br>
From my perspective I know that maintaining a SIP stack
requires *A LOT* of effort, so I understand that a
project can’t maintain two of them.<br>
<br>
I suggest that a working group is created for the
transition and that the first task is to compare the
functionality.
<br>
Last time I checked the functionality *I need* (but
maybe not everyone else) was non-existing in PJSIP so I
could not use it.<br>
It may have changed since then.<br>
<br>
I think the goal has to be to gradually phase out the
ugly code in chan_sip and celebrate the day it’s gone,
but<br>
make sure we don’t leave functionality (and users)
behind and have good guidelines for the transition.<br>
<br>
I still think we should totally rewrite how chan_pjsip
is configured. That concept is very far away from other
SIP implementations.<br>
But that’s my personal opinion from a small cold corner
of the world, using Asterisk in non-PBX ways as large
scale media
<br>
and feature servers.<br>
<br>
Executive summary: Create a working group that maintains
the feature gap, makes sure it’s going away and also
makes sure<br>
that we have enough material that explains the gold that
hides in chan_pjsip!<br>
<br>
/O<br>
-- <br>
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