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Hello,<br>
<br>
run asterisk in debug mode to understand why is sending BYE.<br>
<br>
Cheers,<br>
Daniel<br>
<br>
<div class="moz-cite-prefix">On 07/08/15 16:40, Loic Chabert wrote:<br>
</div>
<blockquote
cite="mid:CADLP+M0Nra9QeE8DC6cmvh-rAwpewFaj=zfSRh31zKuMBUbqyw@mail.gmail.com"
type="cite">
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<div>Hello, <br>
<br>
</div>
I have set on the right place
"route(RTPPROXY")", and now it works for
internal calls and external calls.<br>
</div>
<div>Reason: my request passing througt RTPPROXY
twice ...<br>
</div>
<div><br>
</div>
One last problem:<br>
</div>
- 102 initiate a call to 101<br>
</div>
- 101 refuse call with a 486 response<br>
</div>
- as asterisk dialplan said: launch voicemail app<br>
</div>
- Sounds files has been read from asterisk, but after 5
secondes, session has been cut with a BYE request sent
by Asterisk.<br>
<br>
</div>
Please find in attachement pcap trace file (91.x.x.x is
wan kamailio interface, 10.0.247.197 is lan kamailio
interface, facing to asterisk cluster)<br>
<br>
</div>
Why asterisk send this BYE ? Kamailio does not force him to
send this BYE...<br>
<br>
</div>
Thanks,<br>
</div>
Loic.<br>
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<div class="gmail_extra"><br>
<div class="gmail_quote">2015-08-07 10:34 GMT+02:00
Daniel-Constantin Mierla <span dir="ltr"><<a
moz-do-not-send="true" href="mailto:miconda@gmail.com"
target="_blank">miconda@gmail.com</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000"> Hello,<br>
<br>
look at the sip traffic and see what is in SDP, if you
don't get audio, maybe the other ip is advertised.<br>
<br>
Cheers,<br>
Daniel
<div>
<div class="h5"><br>
<br>
<div>On 07/08/15 09:16, Loic Chabert wrote:<br>
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<blockquote type="cite">
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<div>Hello Daniel, <br>
<br>
</div>
I have changed my rtpproxy by rtpengine.
I have explicitly define public and
private interfaces, and now it work as
expected for external calls (througth
PSTN).<br>
</div>
But for now, after this change, internal
call (like 100 call 101), does not work
any more.<br>
<br>
</div>
I need more investigation to see what append
on my call flow.<br>
<br>
</div>
I will update you asap.<br>
<br>
</div>
Thanks,<br>
</div>
Regards.<br>
<div>
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<div><br>
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<div class="gmail_extra"><br>
<div class="gmail_quote">2015-08-07 9:04 GMT+02:00
Daniel-Constantin Mierla <span dir="ltr"><<a
moz-do-not-send="true"
href="mailto:miconda@gmail.com"
target="_blank">miconda@gmail.com</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0
0 0 .8ex;border-left:1px #ccc
solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000"> Hello,<span><br>
<br>
<div>On 30/07/15 17:38, Loic Chabert
wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">
<div>
<div>Hello everyone,<br>
<br>
</div>
I'm trying put kamailio in front of
asterisk server farm. Fow now, 2
asterisk servers are running and i'm
trying to make some basic calls
between two UACc.<br>
<br>
All asterisk servers has been
ofuscaded from public internet using
<a moz-do-not-send="true"
href="http://10.189.122.0/24"
target="_blank">10.189.122.0/24</a>
network.<br>
</div>
<div>All trafic must be passed
throught asterisk so RTPproxy is
used to (and used for rtp bridging).<br>
</div>
<div>Kamailio and rtpproxy is running
with public IP address, and private
ip address (mhomed=1)<br>
</div>
<div><br>
</div>
<div>But a wired thing append on my
SDP body: c line have two rtpproxy
public addresses concatenate (see my
capture attached).<br>
<br>
</div>
<div>Any reason for this ? Only invite
method from my asterisk contains 2
publics IP addresses concatenated.<br>
</div>
<div><br>
</div>
<div>Does it mean than rtp_manage as
been executed twice ?<br>
<br>
</div>
</div>
</blockquote>
</span> It could be that it was executed
twice. As pointed in another response, look
at what is received on the network and in
the logs.<br>
<br>
You can enable cfgtrace for debugger module
in order to see what actions are executed
from configuration files -- it is good to
spot quickly errors in the logic of config
file.<br>
<br>
Cheers,<br>
Daniel<span><font color="#888888"><br>
<br>
<pre cols="72">--
Daniel-Constantin Mierla
<a moz-do-not-send="true" href="http://twitter.com/#%21/miconda" target="_blank">http://twitter.com/#!/miconda</a> - <a moz-do-not-send="true" href="http://www.linkedin.com/in/miconda" target="_blank">http://www.linkedin.com/in/miconda</a>
Book: SIP Routing With Kamailio - <a moz-do-not-send="true" href="http://www.asipto.com" target="_blank">http://www.asipto.com</a></pre>
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<br>
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</blockquote>
<br>
<pre cols="72">--
Daniel-Constantin Mierla
<a moz-do-not-send="true" href="http://twitter.com/#%21/miconda" target="_blank">http://twitter.com/#!/miconda</a> - <a moz-do-not-send="true" href="http://www.linkedin.com/in/miconda" target="_blank">http://www.linkedin.com/in/miconda</a>
Book: SIP Routing With Kamailio - <a moz-do-not-send="true" href="http://www.asipto.com" target="_blank">http://www.asipto.com</a></pre>
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</blockquote>
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<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>
Book: SIP Routing With Kamailio - <a class="moz-txt-link-freetext" href="http://www.asipto.com">http://www.asipto.com</a></pre>
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