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<tt>Hi guys,<br>
<br>
The normal approach will be to do t_newtran() asap in your script
-> this will take care and absorb retransmissions, but has the
downside of creating the transaction at that moment, so whatever
changes you do later (over the message) will not be recorded into
transaction (if you need them in failure route, for serial
forking).<br>
<br>
Regards,<br>
Bogdan<br>
</tt>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
<br>
On 08/16/2012 11:52 AM, Muhammad Shahzad wrote:
<blockquote
cite="mid:CAFZQphyQamFpCH1A5_fpK0Da5jPj9YQBXYpSGEyHQkQmHMkKOw@mail.gmail.com"
type="cite">I had a similar problem a while ago and i deal with it
in my own way, which was quite simple actually.
<div><br>
</div>
<div>1. First INVITE comes in, i check it does not has to_tag, i
put its call-id in memcache and do normal processing.</div>
<div>2. For subsequent INVITE without to_tag, i check in memcache
if call-id already exists, if yes then send stateless reply
100-Trying and exit. Otherwise treat it as first INVITE.</div>
<div>3. Memcache auto expires each record after 60 seconds, so we
have auto clean up regardless if call was successful or failed.</div>
<div><br>
</div>
<div>If there is any better way, i would be happy to learn it.</div>
<div><br>
</div>
<div>Thank you.</div>
<div><br>
<br>
<div class="gmail_quote">On Thu, Aug 16, 2012 at 7:25 AM, Andrew
Mason <span dir="ltr"><<a moz-do-not-send="true"
href="mailto:andrew.mason@voice.net.au" target="_blank">andrew.mason@voice.net.au</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;">Hi All,<br>
We recently encountered an issue in our datacenter where an
upstream switch<br>
between the CPE and the OpenSIPS server was dropping
packets. Obviously for<br>
VoIP this is not ideal but it did expose an issue in our
configuration.<br>
<br>
If the packet dropped was a '183 Session Progress' or a
'Ringing' then the CPE<br>
would send the same Invite (exact same packet) to which our
server issues a<br>
407 Proxy Auth Required. At which stage the CPE seems to
hang up.<br>
<br>
My question: How can I determine if a call is already in
progress for a<br>
particular Invite ?<br>
<br>
Apologies if my terminology is not correct, please let me
know if anything<br>
does not make sense. I have inherited this system and am
still very much<br>
learning about SIP and OpenSIPS.<br>
<br>
Thanks in advance<br>
Andrew<br>
<br>
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</blockquote>
</div>
<br>
<br clear="all">
<div><br>
</div>
-- <br>
Muhammad Shahzad<br>
-----------------------------------<br>
CISCO Rich Media Communication Specialist (CRMCS)<br>
CISCO Certified Network Associate (CCNA)<br>
Cell: +92 334 422 40 88<br>
MSN: <a moz-do-not-send="true"
href="mailto:shari_786pk@hotmail.com">shari_786pk@hotmail.com</a><br>
Email: <a moz-do-not-send="true"
href="mailto:shaheryarkh@googlemail.com">shaheryarkh@googlemail.com</a><br>
</div>
<pre wrap="">
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</blockquote>
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