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Hi<br>
Below is the comment that written in
chan_sip.c(handle_request_refer) file of asterisk .In RFC also
mentioned that if blind transfer failed call should connect back,
some of phones support this(If received refer) like cisco,polycom
and etc.<br>
<br>
<font color="#ff6666">\par Blind transfers<br>
The transferor provides the transferee<br>
with the transfer targets contact. The signalling between<br>
transferer or transferee should not be cancelled, so the<br>
call is recoverable if the transfer target can not be
reached <br>
by the transferee.<br>
<br>
In this case, Asterisk receives a TRANSFER from<br>
the transferor, thus is the transferee. We should<br>
try to set up a call to the contact provided<br>
and if that fails, re-connect the current session.<br>
If the new call is set up, we issue a hangup.<br>
In this scenario, we are following section 5.2<br>
in the SIP CC Transfer draft. (Transfer without<br>
a GRUU)</font><br>
<br>
<br>
<br>
In asterisk comment is written correct but it is not working.<br>
<br>
Thanks<br>
Nikhil<br>
<br>
On 07/05/2011 09:44 PM, Kevin P. Fleming wrote:
<blockquote cite="mid:4E13385A.8090509@digium.com" type="cite">On
07/05/2011 01:54 AM, Olivier wrote:
<br>
<blockquote type="cite">
<br>
<br>
2011/7/5 Nikhil <<a class="moz-txt-link-abbreviated" href="mailto:d.nikhil@cem-solutions.net">d.nikhil@cem-solutions.net</a>
<br>
<a class="moz-txt-link-rfc2396E" href="mailto:d.nikhil@cem-solutions.net"><mailto:d.nikhil@cem-solutions.net></a>>
<br>
<br>
Hi all
<br>
In asterisk if blind transfer failed ,call is not
connecting back .
<br>
<br>
For Eg:
<br>
A make call to B through asterisk,then B transfer the
call to C.
<br>
If C did not answer the call ,A and B Call should connect
back.
<br>
<br>
IMHO, blind tranfer definition is to NOT connect A and B back
<br>
</blockquote>
<br>
That is correct, and is why it's called a 'blind' transfer; the
transferring party does not know or care what happens to the call
after effecting the transfer.
<br>
<br>
</blockquote>
<br>
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