Giovanni: Yeh, I saw the section regarding sending dtmf after bridge later. Ｉgot too excited once the installation was done and not finish reading the whole thing. Thanks for making things more clear here. I will put your example to the wiki if it's not already there. By the way, is there a place where DTMF is explained in more detail of what are the differences between rfc2833, inband, and info? For my understanding, I think both inband and rfc2833 are transmitted via the RTP correct? and sip info is by the SIP signal? Thanks, Henry Huang US: +1(818)6885508 | 台灣(Taiwan): +886 933847619 Contact Me [image: LinkedIn]<http://www.linkedin.com/profile/view?id=4590380> [image: Facebook] <http://www.facebook.com/profile.php?id=595148342> [image: Twitter] <http://twitter.com/unicsolution> IM [image: Google Talk] red_rain_seven@gmail.com [image: Skype] unicsolution [image: MSN] b_ball_henry@hotmail.com On Sat, Jun 4, 2011 at 1:39 PM, Giovanni Maruzzelli <gmaruzz@gmail.com>wrote:
You need to set the variable 'skype_add_outband_dtmf_also_when_bridged' to true before bridging.
This is my /usr/local/freeswitch/conf/dialplan/default/02_conf_via_skype.xml :
<include> <extension name="conf_via_skype"> <condition field="destination_number" expression="^5004$"> <action application="set" data="skype_add_outband_dtmf_also_when_bridged=true"/> <action application="set" data="ringback=$${us-ring}"/> <action application="sleep" data="500"/> <action application="bridge" data="sofia/${use_profile}/888@conference.freeswitch.org"/> </condition> </extension> </include>
You can copy and paste it in the same file.
You then will edit the file /usr/local/freeswitch/conf/ivr_menus/demo_ivr.xml and substitute the line:
<entry action="menu-exec-app" digits="1" param="bridge sofia/$${domain}/888@conference.freeswitch.org"/>
with the line:
<entry action="menu-exec-app" digits="1" param="transfer 5004 XML default"/>
and it will work as you expect, both with inbound skype and with inbound sip.
-giovanni
On Fri, Jun 3, 2011 at 11:08 PM, Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:
As per the wiki page, and as is clearly written in the console and the logfile, if an incomong skype call is bridged to an outbound call, the dtmf is by default passed only in band (eg: as audio). If you want to have it working with the fs developer conference, create an extension that goes there and add to it the variable that's in the wiki page (can't remember the name, is something like "dtmf-outband-also-when-bridged=true"). Please check the wiki page for the variable name. Reason for this is that the in band dtmf cannot be shut up, and often is detected by the remote party. But in this particular case of the fs developer conference, the remote party is setup to not detect inband in sip (or anyway, it does not detect it). Long story short: create an extension that sets that variable and goes to the fs conference and you'll be all set. Tomorrow I'll post here an example from my dialplan. -giovanni
On 6/3/11, Michael Collins <msc@freeswitch.org> wrote:
Try transferring to a local conference on your machine and see what happens. Watch the fs_cli and see if the DTMFs show up or not. From there we can see what's up...
-MC
On Fri, Jun 3, 2011 at 11:10 AM, Henry Huang <b_ball_henry@hotmail.com>wrote:
But the DTMF press worked very well while I was under the Demo IVR, even when I press "1001" I was able to get connected to extension 1001. It didn't work only after I was transferred to the conference selection.
Henry
On Sat, Jun 4, 2011 at 12:35 AM, Michael Collins <msc@freeswitch.org>wrote:
The monkeys have stopped screaming for everyone. :( I have been looking for an alternative sound file for this but haven't found anything I really like.
Not sure about the DTMF, but from what I understand there have been issues with sending DTMFs from the Skype client. I haven't tried it myself so I will defer to those who have.
-MC
On Fri, Jun 3, 2011 at 9:19 AM, Henry Huang <b_ball_henry@hotmail.com>wrote:
Hi:
The new interactive installation for mod_skypopen is a piece of cake. Made live much easier. I was so excited and tried it out right after the installation was done. But I found a few issues calling the demo-ivr from skyTpe client.
1. The screaming monkey does not scream anymore. 2. The conference that connects to the freeswitch.org doesn't respond to dtmf pressed on the skype client. Ever since the second client is connected to the conference, the conference tell the client it's muted. So I tried push 0 and any other key to see if I get to unmute. But no, non of the key press works.
If anyone have previous experiences on these and fixed them, please do share your methods.
Sound quality with the new virtual sound drive is very good.
Thanks,
Henry Huang US: +1(818)6885508 | 台灣(Taiwan): +886 933847619 Contact Me [image: LinkedIn]<http://www.linkedin.com/profile/view?id=4590380> [image: Facebook] <http://www.facebook.com/profile.php?id=595148342> [image: Twitter] <http://twitter.com/unicsolution> IM [image: Google Talk] red_rain_seven@gmail.com [image: Skype]unicsolution [image: MSN] b_ball_henry@hotmail.com
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Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
-- Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
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