<div dir="ltr"><div><div><div><div><div><div><div><div><div><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><div><div><div><div><div><div><div><div><div><div><br></div></div></div></div></div></div></div></div></div></div></div><div class="gmail_extra"><br><div class="gmail_quote">2015-08-07 10:34 GMT+02:00 Daniel-Constantin Mierla <span dir="ltr"><<a 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>
</div>
<blockquote type="cite">
<div dir="ltr">
<div>
<div>
<div>
<div>
<div>
<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>
<div>
<div>
<div><br>
</div>
</div>
</div>
</div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">2015-08-07 9:04 GMT+02:00
Daniel-Constantin Mierla <span dir="ltr"><<a 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 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 href="http://twitter.com/#%21/miconda" target="_blank">http://twitter.com/#!/miconda</a> - <a href="http://www.linkedin.com/in/miconda" target="_blank">http://www.linkedin.com/in/miconda</a>
Book: SIP Routing With Kamailio - <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a></pre>
</font></span></div>
<br>
_______________________________________________<br>
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
mailing list<br>
<a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a><br>
<a href="http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users" rel="noreferrer" target="_blank">http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users</a><br>
<br>
</blockquote>
</div>
<br>
</div>
</blockquote>
<br>
<pre cols="72">--
Daniel-Constantin Mierla
<a href="http://twitter.com/#!/miconda" target="_blank">http://twitter.com/#!/miconda</a> - <a href="http://www.linkedin.com/in/miconda" target="_blank">http://www.linkedin.com/in/miconda</a>
Book: SIP Routing With Kamailio - <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a></pre>
</div></div></div>
</blockquote></div><br></div>