On Dec 28, 2007 6:22 AM, Kamanashis Roy Shuva <<a href="mailto:kamanashisroy@gmail.com">kamanashisroy@gmail.com</a>> wrote:<br><div class="gmail_quote"><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hi,<br><br> I am expecting someone to discuss this who have at least a little<br>idea about sip actually.</blockquote><div><br><br>You're already discussing this w/ Olle J., who is lead developer and a SIP expert in the Asterisk community :-)
<br><br>I guess the source of the confusion is that sometimes RFC 2833 is used synonymously w/ DTMF, because typically many end-points support only the DTMF portion (0-11) of the RFC. Asterisk supports 0-16 and negotiates that range in the SDP if these events are expected to be conveyed through RTP.
<br><br>With SIP INFO there is no negotiation of the range on what the participating end-points support so all bets are off (that's why using SIP INFO for telephony events is a bad idea).<br><br>--<br>Raj <br><br><br>
</div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><div></div><div class="Wj3C7c"><br>On Dec 28, 2007 5:20 PM, Kamanashis Roy Shuva <
<a href="mailto:kamanashisroy@gmail.com">kamanashisroy@gmail.com</a>> wrote:<br>> Hi,<br>><br>> Let me copy some code from chan_sip.c<br>><br>> I am reading a function body<br>><br>> /*! \brief Receive SIP INFO Message
<br>> \note Doesn't read the duration of the DTMF signal */<br>> static void handle_request_info(struct sip_pvt *p, struct sip_request *req)<br>><br>> ....<br>> ....<br>><br>> if (event == 16) {
<br>> /* send a FLASH event */<br>> struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH, };<br>> ast_queue_frame(p->owner, &f);<br>> if (sipdebug)
<br>> ast_verbose("* DTMF-relay event received: FLASH\n");<br>> } else {<br>><br>> .....<br>> .....<br>><br>> The above code surely works with info message where the message
<br>> body consists of signal value. And that value is matched to 16. This<br>> is enough evidence that asterisk supports flash when it receives<br>> info message.<br>><br>><br>> On Dec 28, 2007 4:41 PM, Johansson Olle E <
<a href="mailto:oej@edvina.net">oej@edvina.net</a>> wrote:<br>> ><br>> > 27 dec 2007 kl. 19.54 skrev Kamanashis Roy Shuva:<br>> ><br>> > > Hi,<br>> > ><br>> > > The patch is not successful adding this feature. I think there is more
<br>> > > to be added. Or the patch should follow a good design. I mean I am not<br>> > > satisfied with this patch anyway.<br>> > ><br>> > > Here I have found the flash support for sip.
<br>> > ><br>> > > <a href="http://www.voip-info.org/wiki/view/SIP+DTMF+Signalling" target="_blank">http://www.voip-info.org/wiki/view/SIP+DTMF+Signalling</a><br>> > ><br>> > ><br>> > This page is describing RFC 2833 DTMF.
<br>><br>> I think you should have a better look. Please recheck ..<br>><br>> ><br>> > > Again, asterisk can accept INFO dtmf flash .. I mean it is in the code<br>> > > that when there is<br>
> > > a info message with "signal=16" it indicates flash.<br>> > ><br>> > > I hope we will be discussing this upto a solution.<br>> ><br>> > Cisco docs indicate that they only support hook flash in the RTP
<br>> > stream. I have not seen any support<br>> > of hook flash in SIP INFO.<br>> ><br>> ><br>> > /O<br>> ><br>> > _______________________________________________<br>> > --Bandwidth and Colocation Provided by
<a href="http://www.api-digital.com--" target="_blank">http://www.api-digital.com--</a><br>> ><br>> > asterisk-dev mailing list<br>> > To UNSUBSCRIBE or update options visit:<br>> > <a href="http://lists.digium.com/mailman/listinfo/asterisk-dev" target="_blank">
http://lists.digium.com/mailman/listinfo/asterisk-dev</a><br>> ><br>><br>><br>><br>> --<br>> -- Thanks<br>><br>> Kamanashis Roy<br>><br><br><br><br>--<br>-- Thanks<br><br>Kamanashis Roy<br><br>
_______________________________________________<br>--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--" target="_blank">http://www.api-digital.com--</a><br><br>asterisk-dev mailing list<br>To UNSUBSCRIBE or update options visit:
<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-dev" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-dev</a><br></div></div></blockquote></div><br>