<html><head><meta http-equiv="Content-Type" content="text/html charset=iso-8859-1"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><br><div><div>10 maj 2013 kl. 10:46 skrev Jaco Kroon <<a href="mailto:jaco@uls.co.za">jaco@uls.co.za</a>>:</div><br class="Apple-interchange-newline"><blockquote type="cite">
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<div class="moz-cite-prefix"><font face="Helvetica, Arial,
sans-serif">Hi Olle,<br>
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On 10/05/2013 10:31, Olle E. Johansson wrote:<br>
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<div>19 apr 2013 kl. 12:34 skrev Jaco Kroon <<a moz-do-not-send="true" href="mailto:jaco@uls.co.za">jaco@uls.co.za</a>>:</div>
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sans-serif">Hi Olle,<br>
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I can confirm your observations. For this reason I've
updated (or initiated the process) to periodically
resolve the SRV records out-of-band and generate an
updated configuration file that contains all of the
hosts. I must further point out that (last I checked)
it gets even worse, if you have an A record with
multiple IPs listed, asterisk will only use one of those
as well.<br>
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I am not sure I like supporting round-robin DNS as the official
recommendation is to use SRV records. The idea with round-robin
DNS is not failover, but to load balance and this works
perfectly well. </div>
<div>After som googling last week, I do realize too many people
use rr DNS so we might have some optional support for it</div>
<div>regardless of my personal feelings. It's not part of the
current project though, my funding is only focused on SRV
support.</div>
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Fine by me. Not my use-case either ... and if you need load
balancing, I'd recommend doing it via SRV by adding multiple RRs
with the same priority, with potentially differing weights anyway.<br>
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<div>I am currently working on fixing Dial() and SRV in SIP.<br>
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This applies to both SIP and IAX/2 (and probably
other protocols that I don't even know about).<br>
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The "other" protocol is out of scope for this project.</div>
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What would it take to include it?<br></div></blockquote>Contact me off-list to discuss it.</div><div><br><blockquote type="cite"><div text="#000000" bgcolor="#FFFFFF">
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Currently my suggestion would be to deal with the
outbound situation in Dialplan(), and to generate
multiple peer configs in the config files, covering each
host individually. Obviously this implies that you have
to track DNS changes external to asterisk. My setup
also utilizes the inbound ACLs for dealing with IAX/2
authorization (so all peers will send IE username=foo,
and [foo] will deny=all,
permit=${ips_from_srv_and_a_rrs}). Still need to
generate multiple peers for outbound cases though.<br>
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That sounds a bit messy to me. I have a more elegant solution
based on the code. I don't know if it works yet, so thank you
for your suggestion, I might have to revert to that if this
fails ;-)</div>
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Not saying it's not messy, just saying it's something that can work
*now* without changes to the code.<br></div></blockquote>Well, it won't give you support for subscribe, publish and quite a lot of other functions where you need SRV load balancing and failover. It will just help call setup. We do have func_srv.c today, but I personally don't feel it's a good long-term solution. A nice hack, but nothing more.</div><div><br><blockquote type="cite"><div text="#000000" bgcolor="#FFFFFF">
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If you'd like to fix this in the code - something which
should probably done, and would be a better solution
than my hack, but which I suspect is going to be rather
complex - I'd be more than willing to help test for you.<br>
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Looking forward to your test results.</div>
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As long as it applies to ast 11 :). What's your time frames?
Trying to decide how much effort to put into my messy idea or
whether I should wait for your patch? I might base a patch for the
"other" protocol off of your work for SIP if it's reasonably
trivial.<br></div></blockquote>The code in pgtips is based on 1.8 but it should not be a lot of work to modify it to 11. I hope to have something finished by the end of the month, as we need to move on to stage 2 which involves some strange IMS-specific stuff.</div><div><br></div><div>/O<br><blockquote type="cite"><div text="#000000" bgcolor="#FFFFFF">
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Kind Regards,<br>
Jaco<br>
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<span><jaco.vcf></span>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --<br><br>asterisk-dev mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-dev">http://lists.digium.com/mailman/listinfo/asterisk-dev</a></blockquote></div><br></body></html>