Most times it's easier to find something in google, than in your own computer :)<br><br><div><span class="gmail_quote">2008/4/25, Eric Wieling <<a href="mailto:eric@fnords.org">eric@fnords.org</a>>:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
In 1.2 it is documented in /path/to/src/asterisk/doc/README.variables,<br> in 1.4 the file is called /path/to/src/asterisk/doc/channelvariables.txt<br> <br> The "doc" directory is the only official source of documentation for<br>
Asterisk that I am aware of. Read it.<br> <br><br> <a href="mailto:bee.beeep@gmail.com">bee.beeep@gmail.com</a> wrote:<br> > Dinesh Nair пишет:<br> >> On Tue, 22 Apr 2008 11:54:41 +0100, Grey Man wrote:<br> >><br>
>>> The best option is to put a SIP Proxy in front of your Asterisk sever<br> >>> and block REFER requests.<br> >>><br> >> or just comment out the block in chan_sip.c which handles the refers.<br>
>><br> >><br> ><br> > Thanks to your answers, but i found more beautiful way to do this -<br> > there is some system variable __TRANSFER_CONTEXT, which defines context<br> > to handle the transfered number, so you can create a new context and<br>
> there you can do anything with transfered call - i just hang it up.<br> ><br> > It's really strange that this is in fact undocumented function - you can<br> > find it only in comments on wiki at <a href="http://voip-info.org">voip-info.org</a>. Man there said that<br>
> he found this variable while hacking source code of asterisk:<br> ><br> > $ grep -R TRANSFER_CONTEXT /usr/src/asterisk-1.2.15/<br> > /usr/src/asterisk-1.2.15/channels/chan_sip.c: *transfercontext =<br> > pbx_builtin_getvar_helper(sip_pvt->owner, "TRANSFER_CONTEXT");<br>
> /usr/src/asterisk-1.2.15/doc/README.variables:${TRANSFER_CONTEXT}<br> > Context for transferred calls<br> > /usr/src/asterisk-1.2.15/ChangeLog: * channels/chan_sip.c: chan_sip did<br> > not use the TRANSFER_CONTEXT<br>
> /usr/src/asterisk-1.2.15/res/res_features.c: if<br> > (!(transferer_real_context = pbx_builtin_getvar_helper(transferee,<br> > "TRANSFER_CONTEXT")) &&<br> > /usr/src/asterisk-1.2.15/res/res_features.c: !(transferer_real_context =<br>
> pbx_builtin_getvar_helper(transferer, "TRANSFER_CONTEXT"))) {<br> > /usr/src/asterisk-1.2.15/res/res_features.c: if<br> > (!(transferer_real_context=pbx_builtin_getvar_helper(transferee,<br> > "TRANSFER_CONTEXT")) &&<br>
> /usr/src/asterisk-1.2.15/res/res_features.c:<br> > !(transferer_real_context=pbx_builtin_getvar_helper(transferer,<br> > "TRANSFER_CONTEXT"))) {<br> ><br> ><br> <br>> _______________________________________________<br>
> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --<br> ><br> > asterisk-users mailing list<br> > To UNSUBSCRIBE or update options visit:<br> > <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
<br> <br>--<br> Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,<br> T-1, PRI, Frame Relay, Linux, and network design. Based near<br> Birmingham, AL. Now accepting clients worldwide.<br> <br><br>
<br> _______________________________________________<br> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --<br> <br> asterisk-users mailing list<br> To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a></blockquote></div><br>