Time to get a full debug log with sip trace and put on pastebin. Be sure to use "FreeSWITCH Log" as the syntax highlighting.<div><br></div><div>-MC<br><br><div class="gmail_quote">On Sun, May 15, 2011 at 12:43 PM, Kamen <span dir="ltr"><<a href="mailto:sireeps@gmail.com">sireeps@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">I spent a few more hours and I figured a few more things. It looks like<br>
there was a lot of excessive configuration for the purpose. I think what<br>
should work (although it still does not) is as follows. No need for any java<br>
scripts in this new version.<br>
<br>
I set the public.xml extension:<br>
<br>
<extension name="extname"><br>
<condition field="caller_id_number" expression="^(9051212121)$"><br>
<action application="set" data="domain_name=$${domain}"/><br>
<action application="transfer" data="3472 XML default"/><br>
</condition><br>
</extension><br>
<br>
And the default.xml goes like this:<br>
<br>
<extension name="extname"><br>
<condition field="destination_number" expression="3472"><br>
<action application="set" data="ringback=$${us-ring}"/><br>
<action application="set" data="transfer_ringback=$${us-ring}"/><br>
<action application="set" data="hangup_after_bridge=true"/><br>
<action application="bridge"<br>
data="sofia/gateway/mysipprovider/15191212121"/><br>
</condition><br>
</extension><br>
<br>
So this is way simpler, although the result is the same - the channel is<br>
opened (2011-05-15 5:33:45.687500 [NOTICE] switch_channel.c:812 New Channel<br>
sofia/external/15191212121 [430cf1e6-0876-4cc7-ba87-9c962619b341]), but I<br>
still get paused for 10 seconds. After which I get short busy signals.<br>
<br>
What I noticed, and that seems to be a clue, if I do not hang up on calling<br>
side and continue with the busy signal, the connection gets through to the<br>
destination number and it shows the caller ID and rings shortly. The way I<br>
see it it is a break through! I never got any rings before on the<br>
destination side.<br>
<br>
Anyway, so the question I have now, why the heck it seems not calling on<br>
destination right away as soon as the channel is opened but pauses until<br>
disconnected? And then actually still calls, but only a short ring, enough<br>
to display a caller ID.<br>
<br>
If anyone had similar problem solved, please, let me know. I would really<br>
appreciate it.<br>
<br>
Regards,<br>
<br>
Kamen<br>
<font color="#888888"><br>
<br>
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