I don't see the siptrace in that log?<div><br></div><div>-Steve</div><div><br></div><div><br><div class="gmail_quote">On 9 June 2011 08:35, Alessandro <span dir="ltr"><<a href="mailto:a.luppi@seletech.com">a.luppi@seletech.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div text="#000000" bgcolor="#ffffff">
Hi,<br>
<br>
the url is: <a href="http://pastebin.freeswitch.org/16463" target="_blank">http://pastebin.freeswitch.org/16463</a><br>
i made a call from phone1 to phone2, the called party refused the
call with code 603. FS received the status 603 form the called
(softphone 2) party. Than FS sent to the calling party (softphone 1)
the message 200 and bye.<br>
<br>
This is the resume of the log:<div class="im"><br>
<br>
<div>1000@localnet_ip FS(ip:localnet_ip)
1001<span style="font-family:arial;font-size:small">@localnet_ip</span></div>
<div><br>
</div>
<div><span style="font-family:monospace;font-size:medium">
<pre>INVITE ---------->
INVITE --------------->
<-------------- trying <--------------------trying
<------------------ 603
<span style="font-family:arial;font-size:small"><span style="font-family:monospace;font-size:medium"><pre><-------------- 200</pre></span></span>ACK------------------>
<--------------------BYE</pre>
</span></div>
<br>
<br>
</div><div class="im"><i>You said you had voicemail before... you can't send 603 back to
the client and continue to voicemail because the 603 terminates
the call.</i><br>
<br></div>
When the called party terminates the call before answering, the
calling party receive e registered message like "The phone called is
not available, leave a message ...". Than i found the registered
message in freeswitch. (I'm using fusion-pbx)<br>
<br>
<br>
Thanks<br>
<br>
Regards<br>
<br>
Alessandro<br>
<br>
<br>
Il 08/06/2011 21:55, Steven Ayre ha scritto:
<div><div></div><div class="h5"><blockquote type="cite">
<blockquote style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204, 204, 204);padding-left:1ex" class="gmail_quote"> Question 1:<br>
i'm developing a custom client sip with pjsip. This client when
receive a call that can't be accepted respond with status 603. I
think that freeswitch filter this status.<br>
</blockquote>
<br>
603 gets treated fine for me. I think we need to see more
information - can you put a debug level log of the call with
siptrace enabled (sofia global siptrace on) on pastebin (<a href="http://pastebin.freeswitch.org/" target="_blank">http://pastebin.freeswitch.org/</a>)
and then post the url here?<br>
<br>
Chances are you're doing something in the dialplan that's
answering the call, either before or after the failed bleg.<br>
<br>
You said you had voicemail before... you can't send 603 back to
the client and continue to voicemail because the 603 terminates
the call.<br>
<span style="font-family:arial;font-size:small"><span style="font-family:monospace;font-size:medium"></span></span>
<div> </div>
<blockquote style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204, 204, 204);padding-left:1ex" class="gmail_quote">
Question:2<br>
</blockquote>
<blockquote style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204, 204, 204);padding-left:1ex" class="gmail_quote">It's possible a custom Header pass trough in
status response like trying or session in progress? I'm able to
use custom header only on invite adding to invite a header with
name like X-myheader.
Any suggestion? <br>
</blockquote>
<div><br>
Yes, you can for 180/183, with the sip_ph_X- prefix. That puts
the header on any provisional response.<br>
<a href="http://wiki.freeswitch.org/wiki/Sofia#Adding_Response_Headers" target="_blank">http://wiki.freeswitch.org/wiki/Sofia#Adding_Response_Headers</a><br>
<br>
For example:<br>
<action application="set" data="sip_p_X-myheader=value"/><br>
<br>
AFAIK you won't be able to do the same for a 100 Trying since
Sofia doesn't let FS do any handling at the required point. But
have a try anyway just to be sure.<br>
<br>
</div>
-Steve<br>
<br>
<br>
<div class="gmail_quote">On 8 June 2011 20:14, Alessandro <span dir="ltr"><<a href="mailto:a.luppi@seletech.com" target="_blank">a.luppi@seletech.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204, 204, 204);padding-left:1ex">
<div text="#000000" bgcolor="#ffffff"> Hi,<br>
<br>
I have two questions about FS:<br>
<br>
Question 1:<br>
i'm developing a custom client sip with pjsip. This client
when receive a call that can't be accepted respond with
status 603. I think that freeswitch filter this status.<br>
This is an example of desired behaviour:<br>
<span style="font-family:arial;font-size:small">
<div>1000@localnet_ip
FS(ip:localnet_ip)
1001<span style="font-family:arial;font-size:small">@localnet_ip</span></div>
<div><br>
</div>
<div><span style="font-family:monospace;font-size:medium">
<pre>INVITE ---------->
INVITE --------------->
<-------------- trying <--------------------trying
<-------------- 603 <------------------ 603
ACK ------------> ACK------------------>
</pre>
<pre><span style="white-space:normal"><pre><span style="font-family:monospace;font-size:medium"></span><span style="font-family:arial;font-size:small;white-space:normal"><div>
The current behaviour of FS is:</div><div>
</div></span></pre></span></pre>
</span>
<div>1000@localnet_ip
FS(ip:localnet_ip)
1001<span style="font-family:arial;font-size:small">@localnet_ip</span></div>
<div><br>
</div>
<div><span style="font-family:monospace;font-size:medium">
<pre>INVITE ---------->
INVITE --------------->
<-------------- trying <--------------------trying
<------------------ 603
<span style="font-family:arial;font-size:small"><span style="font-family:monospace;font-size:medium"><pre><-------------- 200</pre></span></span>ACK------------------>
<--------------------BYE
I'd like to avoid the current behaviour. It's possible a kind of message status path trough?
If the called party terminate the call before answering, FS send always to the calling partner 200 and BYE.
First thought was related to the voice-mail. Now voice-mail is disabled but the behaviour is the same.
Question:2
It's possible a custom Header pass trough in status response like trying or session in progress?
I'm able to use custom header only on invite adding to invite a header with name like X-myheader.
Any suggestion?
Thanks
Good Evening
</pre>
</span></div>
</div>
</span>Alessandro<br>
<pre cols="72">--
Ing. Alessandro Luppi
Software development
Seletech srl
Via Collodi 8, 20052 Monza (MI) - Italy
Tel: +39.039.5962000 - Fax: +39.039.9716905
email: <a href="mailto:a.luppi@seletech.com" target="_blank">a.luppi@seletech.com</a> - Web: <a href="http://www.seletech.com" target="_blank">www.seletech.com</a> or <a href="http://www.seletech.eu" target="_blank">www.seletech.eu</a>
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<pre cols="72">--
Ing. Alessandro Luppi
Software development
Seletech srl
Via Collodi 8, 20052 Monza (MI) - Italy
Tel: +39.039.5962000 - Fax: +39.039.9716905
email: <a href="mailto:a.luppi@seletech.com" target="_blank">a.luppi@seletech.com</a> - Web: <a href="http://www.seletech.com" target="_blank">www.seletech.com</a> or <a href="http://www.seletech.eu" target="_blank">www.seletech.eu</a>
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