mod_skypopen is awesome issue with demo-ivr conference
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
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
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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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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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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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-- Sent from my mobile device Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618
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
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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-- Sent from my mobile device
Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
-- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618
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
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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-- Sent from my mobile device
Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
-- Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
One thing I didn't see it covered in the wiki. Is the skype client we use for the mod_skypopen having the risk of being used as a "supernode"? If so, is there ways to disable it? Thanks, Henry 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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-- Sent from my mobile device
Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
-- Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
If you're behind a nat or a firewall you can't be supernode. If you're directly on the internet (eg: you have a real routable IP number) on the machine, and you are not behind a firewall (eg: you have all incoming ports open) then you *may* become a supernode. Btw: google is your friend, those are informations easily available with a search for "skype supernode". -giovanni On 6/5/11, Henry Huang <b_ball_henry@hotmail.com> wrote:
One thing I didn't see it covered in the wiki. Is the skype client we use for the mod_skypopen having the risk of being used as a "supernode"? If so, is there ways to disable it?
Thanks,
Henry
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 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
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-- Sent from my mobile device
Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
-- Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Sent from my mobile device Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618
I think you might still be able to be a supernode behind nat if your router uses upnp... Not sure though. Afaik the option to disable being a supernode was added on Windows in v3... No such option on Linux yet (v2). Steve on iPhone On 5 Jun 2011, at 07:21, Giovanni Maruzzelli <gmaruzz@celliax.org> wrote:
If you're behind a nat or a firewall you can't be supernode. If you're directly on the internet (eg: you have a real routable IP number) on the machine, and you are not behind a firewall (eg: you have all incoming ports open) then you *may* become a supernode. Btw: google is your friend, those are informations easily available with a search for "skype supernode". -giovanni
On 6/5/11, Henry Huang <b_ball_henry@hotmail.com> wrote:
One thing I didn't see it covered in the wiki. Is the skype client we use for the mod_skypopen having the risk of being used as a "supernode"? If so, is there ways to disable it?
Thanks,
Henry
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 >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Sent from my mobile device
Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
-- Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Sent from my mobile device
Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Sorry, I didn't make it clear. I did search google and found "general" information of skype being behind NAT or firewall. But I was looking for something that is capable of turning that feature off. So I tossed the questions here hoping people having done integration with Skype would know better. I also thought of some feature that I am not sure if I should posted here or open a new thread. I was thinking of integrating the mod_skypopen with mod_digaling or some other sort so that when the skype client receive messages, we can transfer it to a designated XMPP client with a 2 way text design in mind. Thanks, Henry On Sun, Jun 5, 2011 at 2:21 PM, Giovanni Maruzzelli <gmaruzz@celliax.org>wrote:
If you're behind a nat or a firewall you can't be supernode. If you're directly on the internet (eg: you have a real routable IP number) on the machine, and you are not behind a firewall (eg: you have all incoming ports open) then you *may* become a supernode. Btw: google is your friend, those are informations easily available with a search for "skype supernode". -giovanni
On 6/5/11, Henry Huang <b_ball_henry@hotmail.com> wrote:
One thing I didn't see it covered in the wiki. Is the skype client we use for the mod_skypopen having the risk of being used as a "supernode"? If so, is there ways to disable it?
Thanks,
Henry
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 >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Sent from my mobile device
Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
-- Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Sent from my mobile device
Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
:) again, read the wiki page. Messages use the CHAT standard freeswitch interface, same used by dingaling ans sofia. So, you can simply make an application that routes messages between skype, google and sip, eg, using the ESL. -giovanni On 6/5/11, Henry Huang <b_ball_henry@hotmail.com> wrote:
Sorry, I didn't make it clear. I did search google and found "general" information of skype being behind NAT or firewall. But I was looking for something that is capable of turning that feature off. So I tossed the questions here hoping people having done integration with Skype would know better.
I also thought of some feature that I am not sure if I should posted here or open a new thread. I was thinking of integrating the mod_skypopen with mod_digaling or some other sort so that when the skype client receive messages, we can transfer it to a designated XMPP client with a 2 way text design in mind.
Thanks,
Henry
On Sun, Jun 5, 2011 at 2:21 PM, Giovanni Maruzzelli <gmaruzz@celliax.org>wrote:
If you're behind a nat or a firewall you can't be supernode. If you're directly on the internet (eg: you have a real routable IP number) on the machine, and you are not behind a firewall (eg: you have all incoming ports open) then you *may* become a supernode. Btw: google is your friend, those are informations easily available with a search for "skype supernode". -giovanni
On 6/5/11, Henry Huang <b_ball_henry@hotmail.com> wrote:
One thing I didn't see it covered in the wiki. Is the skype client we use for the mod_skypopen having the risk of being used as a "supernode"? If so, is there ways to disable it?
Thanks,
Henry
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 >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users@lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
-- Sent from my mobile device
Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
-- Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Sent from my mobile device
Sincerely,
Giovanni Maruzzelli Cell : +39-347-2665618
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Sent from my mobile device Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618
participants (5)
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Giovanni Maruzzelli
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Giovanni Maruzzelli
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Henry Huang
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Michael Collins
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Steven Ayre