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<div class="moz-cite-prefix">On 28/11/2016 14:44, Anthony Messina
wrote:<br>
</div>
<blockquote
cite="mid:20161128074449.Horde.ahnYE_62gR0gB5n_IpVH8N8@messinet.com"
type="cite">I meant that before 4.4.4, I was building from the 4.4
branch will all commits in that branch up until 81df84b . git diff
81df84b 4.4 should show the difference, I just can't figure out
what part isn't working.
<br>
<br>
<br>
I am not using t_suspend()/t_continue() or other async processing
functions</blockquote>
Then I don't recall a change that could affect this behaviour.<br>
<br>
<blockquote
cite="mid:20161128074449.Horde.ahnYE_62gR0gB5n_IpVH8N8@messinet.com"
type="cite"> and I do call route[NATMANAGE] after route[RELAY]
from branch_route[MANAGE_BRANCH].
<br>
</blockquote>
<br>
But I don't really get how you do the above stuff. Maybe you can
paste here the relevant part of config with route RELAY and branch
route....<br>
<br>
Cheers,<br>
Daniel<br>
<br>
<blockquote
cite="mid:20161128074449.Horde.ahnYE_62gR0gB5n_IpVH8N8@messinet.com"
type="cite">
<br>
<br>
Quoting Daniel-Constantin Mierla <a class="moz-txt-link-rfc2396E" href="mailto:miconda@gmail.com"><miconda@gmail.com></a>:
<br>
<br>
<blockquote type="cite">The commit 81df84b is related to tls, so
there should be no relation
<br>
with nat functions.
<br>
<br>
Are you using t_suspend()/t_continue() or other async processing
functions?
<br>
<br>
Anyhow, whatever changes you want to be specific for each
outgoing
<br>
request must be done in a branch_route. request_route changes
are
<br>
visible to all changes and if you do something again in
branch_route or
<br>
failure_route, then you will see duplicated operations.
<br>
<br>
Cheers,
<br>
Daniel
<br>
<br>
<br>
On 28/11/2016 09:02, Anthony Messina wrote:
<br>
<blockquote type="cite">Prior to the upgrade to Kamailio 4.4.4,
I was using 4.4.3 with updates
<br>
through 81df84b from the 4.4 branch and the following
fix_nated_sdp
<br>
call worked through the initial INVITE and when a voicemail
branch was
<br>
appended from the failure_route.
<br>
<br>
route[NATMANAGE] {
<br>
...
<br>
<br>
# RTPEngine is not needed but if STUN addresses are being used
on our
<br>
# internal network, rewrite the addresses to the internal
source address
<br>
if(!isbflagset(FLB_RTPENGINE) &&
has_body("application/sdp")) {
<br>
if(dst_ip==10.1.1.2 &&
<br>
compare_pure_ips($sel(contact.uri.host),
"<EXTERNAL_IP>")) {
<br>
fix_nated_sdp("10");
<br>
}
<br>
}
<br>
<br>
...
<br>
}
<br>
<br>
Since the upgrade to 4.4.4, when the original INVITE fails
over to
<br>
voicemail (branch appended), fix_nated_sdp creates the SDP
below,
<br>
doubling the replaced IP addresses. Asterisk replies with
488, as the
<br>
media address is completely invalid. I need something like
the above
<br>
(which worked for the request and the reply) but can't figure
out what
<br>
the problem is.
<br>
<br>
v=0
<br>
o=Zoiper 0 0 IN IP4 10.1.1.18210.1.1.182
<br>
s=Zoiper
<br>
c=IN IP4 10.1.1.18210.1.1.182
<br>
t=0 0
<br>
m=audio 63732 RTP/SAVP 9 3 0 97 101
<br>
a=rtpmap:9 G722/8000
<br>
a=rtpmap:3 GSM/8000
<br>
a=rtpmap:0 PCMU/8000
<br>
a=rtpmap:97 iLBC/8000
<br>
a=fmtp:97 mode=30
<br>
a=rtpmap:101 telephone-event/8000
<br>
a=fmtp:101 0-16
<br>
a=sendrecv
<br>
a=crypto:5 AES_256_CM_HMAC_SHA1_80
<br>
inline:n2C0BcdXOr32BPTEE8pfHII9mamAK566xMNvQIDIGKI2LPOLdNDOcpFdtqu5DQ==
<br>
a=crypto:6 AES_256_CM_HMAC_SHA1_32
<br>
inline:n2C0BcdXOr32BPTEE8pfHII9mamAK566xMNvQIDIGKI2LPOLdNDOcpFdtqu5DQ==
<br>
a=crypto:3 AES_192_CM_HMAC_SHA1_80
<br>
inline:n2C0BcdXOr32BPTEE8pfHII9mamAK566xMNvQIDIGKI2LPOLdNA=
<br>
a=crypto:4 AES_192_CM_HMAC_SHA1_32
<br>
inline:n2C0BcdXOr32BPTEE8pfHII9mamAK566xMNvQIDIGKI2LPOLdNA=
<br>
a=crypto:1 AES_CM_128_HMAC_SHA1_80
<br>
inline:n2C0BcdXOr32BPTEE8pfHII9mamAK566xMNvQIDI
<br>
a=crypto:2 AES_CM_128_HMAC_SHA1_32
<br>
inline:n2C0BcdXOr32BPTEE8pfHII9mamAK566xMNvQIDI
<br>
a=oldmediaip:<EXTERNAL_IP>
<br>
a=oldmediaip:<EXTERNAL_IP>
<br>
a=oldmediaip:<EXTERNAL_IP>
<br>
a=oldmediaip:<EXTERNAL_IP>
<br>
<br>
<br>
<br>
<br>
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<br>
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<br>
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<br>
</blockquote>
<br>
--
<br>
Daniel-Constantin Mierla
<br>
<a class="moz-txt-link-freetext" href="http://twitter.com/#!/miconda">http://twitter.com/#!/miconda</a> -
<a class="moz-txt-link-freetext" href="http://www.linkedin.com/in/miconda">http://www.linkedin.com/in/miconda</a>
<br>
Kamailio Advanced Training, Berlin, Nov 28-30, 2016 -
<a class="moz-txt-link-freetext" href="http://www.asipto.com">http://www.asipto.com</a>
<br>
</blockquote>
<br>
<br>
<br>
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<pre wrap="">_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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</pre>
</blockquote>
<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla
<a class="moz-txt-link-freetext" href="http://twitter.com/#!/miconda">http://twitter.com/#!/miconda</a> - <a class="moz-txt-link-freetext" href="http://www.linkedin.com/in/miconda">http://www.linkedin.com/in/miconda</a>
Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - <a class="moz-txt-link-freetext" href="http://www.asipto.com">http://www.asipto.com</a></pre>
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