you would need to provide the console output of FreeSWITCH of the entire call.<br>with TPORT_LOG=1 env var set and console loglevel debug (press f8)<br><br><br><div class="gmail_quote">On Fri, Feb 6, 2009 at 2:57 AM, Ankit Gandhi <span dir="ltr"><<a href="mailto:ahgindia308@gmail.com">ahgindia308@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>
Hello,<br>
I want my caller to detect hangup cause as 34 so that he can try next<br>
provider according to lcr.<br>
Here is my setup.<br>
Caller -> switch (fs) -> Terminator.<br>
Now when terminator sends "503 Service Unavailable", I want to override this<br>
cause, so that the caller gets hangup case 34. (According to the terminator<br>
he is sending hangup cause 34 from his side, but in freeswitch we are<br>
getting hangup cause 41 for that call and the same hangup cause on caller<br>
side).<br>
When I tried asterisk as caller, I get hangup cause 34 in that case. But<br>
when I tried freeswitch as caller, then we are getting hangup cause 41, the<br>
same as we are getting in switch (fs).<br>
>From the switch, I tried one of this condition through javascript to<br>
override the hangup cause before sending to caller:<br>
-> session.execute("respond","503");<br>
-> session.execute("hangup","NORMAL_CIRCUIT_CONGESTION");<br>
-> session.execute("hangup","34");<br>
-> session.hangup(34);<br>
In all the above cases, asterisk properly detects the hangup cause 34, but<br>
freeswitch does not detect that. It detects the same hangup cause 41 for the<br>
call. Other callers also get the same hangup cause 41 for such calls.<br>
How can I override this cause, so that the caller gets hangup cause 34 in<br>
such cases?<br>
<br>
Here is the sip trace, on the caller side returned through switch (fs).<br>
ss = switch<br>
cc = caller<br>
--------------------------------------------------------------------<br>
U ss.ss.ss.ss:5060 -> <a href="http://cc.cc.cc.cc:5080" target="_blank">cc.cc.cc.cc:5080</a><br>
SIP/2.0 503 Service Unavailable.<br>
Via: SIP/2.0/UDP<br>
cc.cc.cc.cc:5080;rport=5080;branch=z9hG4bK8HvDU4Z88KU5m;received=122.169.29.122.<br>
From: "654321" <<a href="mailto:sip%3A654321@cc.cc.cc.cc">sip:654321@cc.cc.cc.cc</a>>;tag=8p7aF18aN0mSj.<br>
To: <sip:123456@ss.ss.ss.ss>;tag=Nj8ypjDyvKUXe.<br>
Call-ID: fd1ab774-6ec5-122c-9fac-001cc086141d.<br>
CSeq: 110835048 INVITE.<br>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked.<br>
Accept: application/sdp.<br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,<br>
REFER, UPDATE, REGISTER, INFO, PUBLISH.<br>
Supported: timer, precondition, path, replaces.<br>
Allow-Events: talk, presence, dialog, call-info, sla,<br>
include-session-description, presence.winfo, message-summary, refer.<br>
Reason: Q.850;cause=34;text="NORMAL_CIRCUIT_CONGESTION".<br>
Content-Length: 0.<br>
-----------------------------------------------------------------------<br>
<br>
<br>
--<br>
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</blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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