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In fact, after entering in Asterisk for the first time, my call is
redirected to an other component of my system. This other equiment
redirect the same call to Asterisk a second time.<br>
<br>
It is something like this (it's an IMS architecture) :<br>
<br>
Softphone A --> Equiment --> Asterisk --> Equiment -->
Asterisk --> Equipment --> Softphone B<br>
<br>
During my first passage in Asterisk, it sends an Invite with "Unknow"
in field From. And for my second passage, it have to deny the call
because of the sender identity which is "Unknow".<br>
<br>
Both of my softphones are not registered on Asterisk but on the
equiment you can see before.<br>
<br>
Regards<br>
<br>
--<br>
Rémi Druilhe<br>
<br>
Stefan Gofferje a écrit :
<blockquote cite="mid:48D76A0D.6060905@gofferje.homelinux.org"
type="cite">
<pre wrap="">Hi,
<a class="moz-txt-link-abbreviated" href="mailto:remi.druilhe@orange-ftgroup.com">remi.druilhe@orange-ftgroup.com</a> schrieb:
</pre>
<blockquote type="cite">
<pre wrap="">I have done what you told me to do, but nothing changed. Always the same
problem.
</pre>
</blockquote>
<pre wrap=""><!---->
If I understand your dialplan right, your * is still calling itself via
SIP, right?
This is what is called a loop. You should review your dialplan and
replace all dial(SIP/...@samebox) by goto(respective_context,exten,pri).
Or are you trying to call SIP clients which are registered to the box?
In that case you don't dial(SIP/exten@box) but dial(SIP/accountname)
while accountname is what stands in [] for that client in your sip.conf.
Terve,
Stefan
</pre>
</blockquote>
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