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[Freeswitch-users] no RTP send during Voice Mail recording

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anthony.minessale at g...
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PostPosted: Sat Apr 25, 2009 6:25 pm    Post subject: [Freeswitch-users] no RTP send during Voice Mail recording Reply with quote

more efficient fix should be in tree

i just have to init the buffer one since it never changes.


On Sat, Apr 25, 2009 at 3:56 PM, kokoska rokoska <kokoska.rokoska@post.cz (kokoska.rokoska@post.cz)> wrote:
Quote:

Thank you very much, Dave, for your help!

Mentioned modification did the trick and all works as I wish Smile Thank you!

BTW: When I look into the pcap files for RTP stream, I see some strange
timing for outgoing RTP from FreeSWITCH. Every even packet is sent after
24 ms and every odd packet at 12 ms...
"ptime" is on both sides set to 20 ms, incomming stream has nearly no
jitter, server is on real HW (no virtualization) and almost idle (the
most consuming process is "htop" Smile
Is it normal or should I investigate what is wrong?

Best regards,

kokoska.rokoska


David Knell napsal(a):

Quote:
Add something like
memset(write_buf, 0, SWITCH_RECOMMENDED_BUFFER_SIZE);
after
char write_buf[SWITCH_RECOMMENDED_BUFFER_SIZE];
in switch_ivr_play_say.c (line 395)

--Dave

Quote:
Thank you very much, dujinfang, for your help!

When I use
<action application="set" data="record_waste_resources=true"/>

the FreeSWITCH really sends back RTP stream during recording, but
instead of (faked) silence it is full of completely regular load noise Smile
I have tested it with different devices (Linskys, Snom, FritzBOX,
Nokia...) with the same result (even pcap files looks similar).

Dialplan snipped looks like:
<action application='answer'/>
<action application='playback' data='silence_stream://1000'/>
<action application='set' data='record_waste_resources=true'/>
<action application='voicemail' data='context $${domain} number'/>
<action application='hangup'/>

Do you (or anybody else Smile know what I'm doing wrong?

Thanks once more, dujinfang, for your help!

Best regards,

kokoska.rokoska



dujinfang napsal(a):
Quote:
I haven't tested but I guess it's just like other variables and I
documented to here:

http://wiki.freeswitch.org/wiki/Channel_Variables#record_waste_resources


On Apr 25, 2009, at 11:56 PM, kokoska rokoska wrote:

Quote:
Thank you very much, Anthony, for such fast solution!

May I ask you - How should I activate this feature?
I have tried to "grep" through sources for new NDLB variable but I
didn't find one...

Best regards,

kokoska.rokoska

Anthony Minessale napsal(a):
Quote:
sigh,

see r13144


On Fri, Apr 24, 2009 at 3:43 AM, kokoska rokoska
<kokoska.rokoska@post.cz (kokoska.rokoska@post.cz) <mailto:kokoska.rokoska@post.cz (kokoska.rokoska@post.cz)>> wrote:




   seven napsal(a):
Quote:
You are not alone, I vote 1.

And there's a similer variable in conference:

      <!--Can be | delim of waste|mute|deaf waste will always
transmit data to each channel
          even during silence -->
      <!--<param name="member-flags" value="waste"/>-->

   Thank you very much, seven, for your support Smile

   Best regards,

   kokoska.rokoska


   _______________________________________________
   Freeswitch-users mailing list
   Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
   <mailto:Freeswitch-users@lists.freeswitch.org (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




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
<mailto:MSN%3Aanthony_minessale@hotmail.com ([email]MSN%253Aanthony_minessale@hotmail.com[/email])>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
<mailto:PAYPAL%3Aanthony.minessale@gmail.com ([email]PAYPAL%253Aanthony.minessale@gmail.com[/email])>
IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
<mailto:sip%3A888@conference.freeswitch.org ([email]sip%253A888@conference.freeswitch.org[/email])>
iax:guest@conference.freeswitch.org/888
<http://iax:guest@conference.freeswitch.org/888>
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org ([email]googletalk%253Aconf%252B888@conference.freeswitch.org[/email])>
pstn:213-799-1400


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http://www.freeswitch.org





--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400
Back to top
dujinfang at gmail.com
Guest





PostPosted: Sun Apr 26, 2009 7:38 am    Post subject: [Freeswitch-users] no RTP send during Voice Mail recording Reply with quote

Almost perfect. But I think generate CN fake audio is better than absolute silence. Like bridge_generate_comfort_noise does.




On Apr 26, 2009, at 7:15 AM, Anthony Minessale wrote:
Quote:
more efficient fix should be in tree

i just have to init the buffer one since it never changes.


On Sat, Apr 25, 2009 at 3:56 PM, kokoska rokoska <kokoska.rokoska@post.cz (kokoska.rokoska@post.cz)> wrote:
Quote:

Thank you very much, Dave, for your help!

Mentioned modification did the trick and all works as I wish Smile Thank you!

BTW: When I look into the pcap files for RTP stream, I see some strange
timing for outgoing RTP from FreeSWITCH. Every even packet is sent after
24 ms and every odd packet at 12 ms...
"ptime" is on both sides set to 20 ms, incomming stream has nearly no
jitter, server is on real HW (no virtualization) and almost idle (the
most consuming process is "htop" Smile
Is it normal or should I investigate what is wrong?

Best regards,

kokoska.rokoska


David Knell napsal(a):

Quote:
Add something like
memset(write_buf, 0, SWITCH_RECOMMENDED_BUFFER_SIZE);
after
char write_buf[SWITCH_RECOMMENDED_BUFFER_SIZE];
in switch_ivr_play_say.c (line 395)

--Dave

Quote:
Thank you very much, dujinfang, for your help!

When I use
<action application="set" data="record_waste_resources=true"/>

the FreeSWITCH really sends back RTP stream during recording, but
instead of (faked) silence it is full of completely regular load noise Smile
I have tested it with different devices (Linskys, Snom, FritzBOX,
Nokia...) with the same result (even pcap files looks similar).

Dialplan snipped looks like:
<action application='answer'/>
<action application='playback' data='silence_stream://1000'/>
<action application='set' data='record_waste_resources=true'/>
<action application='voicemail' data='context $${domain} number'/>
<action application='hangup'/>

Do you (or anybody else Smile know what I'm doing wrong?

Thanks once more, dujinfang, for your help!

Best regards,

kokoska.rokoska



dujinfang napsal(a):
Quote:
I haven't tested but I guess it's just like other variables and I
documented to here:

http://wiki.freeswitch.org/wiki/Channel_Variables#record_waste_resources


On Apr 25, 2009, at 11:56 PM, kokoska rokoska wrote:

Quote:
Thank you very much, Anthony, for such fast solution!

May I ask you - How should I activate this feature?
I have tried to "grep" through sources for new NDLB variable but I
didn't find one...

Best regards,

kokoska.rokoska

Anthony Minessale napsal(a):
Quote:
sigh,

see r13144


On Fri, Apr 24, 2009 at 3:43 AM, kokoska rokoska
<kokoska.rokoska@post.cz (kokoska.rokoska@post.cz) <mailto:kokoska.rokoska@post.cz (kokoska.rokoska@post.cz)>> wrote:




seven napsal(a):
Quote:
You are not alone, I vote 1.

And there's a similer variable in conference:

<!--Can be | delim of waste|mute|deaf waste will always
transmit data to each channel
even during silence -->
<!--<param name="member-flags" value="waste"/>-->

Thank you very much, seven, for your support Smile

Best regards,

kokoska.rokoska


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
<mailto:Freeswitch-users@lists.freeswitch.org (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




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
<mailto:MSN%3Aanthony_minessale@hotmail.com ([email]MSN%253Aanthony_minessale@hotmail.com[/email])>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
<mailto:PAYPAL%3Aanthony.minessale@gmail.com ([email]PAYPAL%253Aanthony.minessale@gmail.com[/email])>
IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
<mailto:sip%3A888@conference.freeswitch.org ([email]sip%253A888@conference.freeswitch.org[/email])>
iax:guest@conference.freeswitch.org/888
<http://iax:guest@conference.freeswitch.org/888>
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org ([email]googletalk%253Aconf%252B888@conference.freeswitch.org[/email])>
pstn:213-799-1400


------------------------------------------------------------------------

_______________________________________________
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Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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_______________________________________________
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400
_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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http://www.freeswitch.org
Back to top
anthony.minessale at g...
Guest





PostPosted: Sun Apr 26, 2009 9:49 am    Post subject: [Freeswitch-users] no RTP send during Voice Mail recording Reply with quote

ok done

set the var to
true / false / <desired silence factor>


On Sun, Apr 26, 2009 at 7:09 AM, dujinfang <dujinfang@gmail.com (dujinfang@gmail.com)> wrote:
Quote:
Almost perfect. But I think generate CN fake audio is better than absolute silence. Like bridge_generate_comfort_noise does.





On Apr 26, 2009, at 7:15 AM, Anthony Minessale wrote:
Quote:
more efficient fix should be in tree

i just have to init the buffer one since it never changes.


On Sat, Apr 25, 2009 at 3:56 PM, kokoska rokoska <kokoska.rokoska@post.cz (kokoska.rokoska@post.cz)> wrote:
Quote:

Thank you very much, Dave, for your help!

Mentioned modification did the trick and all works as I wish Smile Thank you!

BTW: When I look into the pcap files for RTP stream, I see some strange
timing for outgoing RTP from FreeSWITCH. Every even packet is sent after
24 ms and every odd packet at 12 ms...
"ptime" is on both sides set to 20 ms, incomming stream has nearly no
jitter, server is on real HW (no virtualization) and almost idle (the
most consuming process is "htop" Smile
Is it normal or should I investigate what is wrong?

Best regards,

kokoska.rokoska


David Knell napsal(a):

Quote:
Add something like
memset(write_buf, 0, SWITCH_RECOMMENDED_BUFFER_SIZE);
after
char write_buf[SWITCH_RECOMMENDED_BUFFER_SIZE];
in switch_ivr_play_say.c (line 395)

--Dave

Quote:
Thank you very much, dujinfang, for your help!

When I use
<action application="set" data="record_waste_resources=true"/>

the FreeSWITCH really sends back RTP stream during recording, but
instead of (faked) silence it is full of completely regular load noise Smile
I have tested it with different devices (Linskys, Snom, FritzBOX,
Nokia...) with the same result (even pcap files looks similar).

Dialplan snipped looks like:
<action application='answer'/>
<action application='playback' data='silence_stream://1000'/>
<action application='set' data='record_waste_resources=true'/>
<action application='voicemail' data='context $${domain} number'/>
<action application='hangup'/>

Do you (or anybody else Smile know what I'm doing wrong?

Thanks once more, dujinfang, for your help!

Best regards,

kokoska.rokoska



dujinfang napsal(a):
Quote:
I haven't tested but I guess it's just like other variables and I
documented to here:

http://wiki.freeswitch.org/wiki/Channel_Variables#record_waste_resources


On Apr 25, 2009, at 11:56 PM, kokoska rokoska wrote:

Quote:
Thank you very much, Anthony, for such fast solution!

May I ask you - How should I activate this feature?
I have tried to "grep" through sources for new NDLB variable but I
didn't find one...

Best regards,

kokoska.rokoska

Anthony Minessale napsal(a):
Quote:
sigh,

see r13144


On Fri, Apr 24, 2009 at 3:43 AM, kokoska rokoska
<kokoska.rokoska@post.cz (kokoska.rokoska@post.cz) <mailto:kokoska.rokoska@post.cz (kokoska.rokoska@post.cz)>> wrote:




   seven napsal(a):
Quote:
You are not alone, I vote 1.

And there's a similer variable in conference:

      <!--Can be | delim of waste|mute|deaf waste will always
transmit data to each channel
          even during silence -->
      <!--<param name="member-flags" value="waste"/>-->

   Thank you very much, seven, for your support Smile

   Best regards,

   kokoska.rokoska


   _______________________________________________
   Freeswitch-users mailing list
   Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
   <mailto:Freeswitch-users@lists.freeswitch.org (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




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
<mailto:MSN%3Aanthony_minessale@hotmail.com ([email]MSN%253Aanthony_minessale@hotmail.com[/email])>
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
<mailto:PAYPAL%3Aanthony.minessale@gmail.com ([email]PAYPAL%253Aanthony.minessale@gmail.com[/email])>
IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
<mailto:sip%3A888@conference.freeswitch.org ([email]sip%253A888@conference.freeswitch.org[/email])>
iax:guest@conference.freeswitch.org/888
<http://iax:guest@conference.freeswitch.org/888>
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org ([email]googletalk%253Aconf%252B888@conference.freeswitch.org[/email])>
pstn:213-799-1400


------------------------------------------------------------------------

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PostPosted: Sun Apr 26, 2009 10:38 am    Post subject: [Freeswitch-users] no RTP send during Voice Mail recording Reply with quote

Again so quick. Thanks. Changed document: http://wiki.freeswitch.org/wiki/Channel_Variables#record_waste_resources
On Apr 26, 2009, at 10:40 PM, Anthony Minessale wrote:
Quote:
ok done

set the var to
true / false / <desired silence factor>


On Sun, Apr 26, 2009 at 7:09 AM, dujinfang <dujinfang@gmail.com (dujinfang@gmail.com)> wrote:
Quote:
Almost perfect. But I think generate CN fake audio is better than absolute silence. Like bridge_generate_comfort_noise does.





On Apr 26, 2009, at 7:15 AM, Anthony Minessale wrote:
Quote:
more efficient fix should be in tree

i just have to init the buffer one since it never changes.


On Sat, Apr 25, 2009 at 3:56 PM, kokoska rokoska <kokoska.rokoska@post.cz (kokoska.rokoska@post.cz)> wrote:
Quote:

Thank you very much, Dave, for your help!

Mentioned modification did the trick and all works as I wish Smile Thank you!

BTW: When I look into the pcap files for RTP stream, I see some strange
timing for outgoing RTP from FreeSWITCH. Every even packet is sent after
24 ms and every odd packet at 12 ms...
"ptime" is on both sides set to 20 ms, incomming stream has nearly no
jitter, server is on real HW (no virtualization) and almost idle (the
most consuming process is "htop" Smile
Is it normal or should I investigate what is wrong?

Best regards,

kokoska.rokoska


David Knell napsal(a):

Quote:
Add something like
memset(write_buf, 0, SWITCH_RECOMMENDED_BUFFER_SIZE);
after
char write_buf[SWITCH_RECOMMENDED_BUFFER_SIZE];
in switch_ivr_play_say.c (line 395)

--Dave

Quote:
Thank you very much, dujinfang, for your help!

When I use
<action application="set" data="record_waste_resources=true"/>

the FreeSWITCH really sends back RTP stream during recording, but
instead of (faked) silence it is full of completely regular load noise Smile
I have tested it with different devices (Linskys, Snom, FritzBOX,
Nokia...) with the same result (even pcap files looks similar).

Dialplan snipped looks like:
<action application='answer'/>
<action application='playback' data='silence_stream://1000'/>
<action application='set' data='record_waste_resources=true'/>
<action application='voicemail' data='context $${domain} number'/>
<action application='hangup'/>

Do you (or anybody else Smile know what I'm doing wrong?

Thanks once more, dujinfang, for your help!

Best regards,

kokoska.rokoska



dujinfang napsal(a):
Quote:
I haven't tested but I guess it's just like other variables and I
documented to here:

http://wiki.freeswitch.org/wiki/Channel_Variables#record_waste_resources


On Apr 25, 2009, at 11:56 PM, kokoska rokoska wrote:

Quote:
Thank you very much, Anthony, for such fast solution!

May I ask you - How should I activate this feature?
I have tried to "grep" through sources for new NDLB variable but I
didn't find one...

Best regards,

kokoska.rokoska

Anthony Minessale napsal(a):
Quote:
sigh,

see r13144


On Fri, Apr 24, 2009 at 3:43 AM, kokoska rokoska
<kokoska.rokoska@post.cz (kokoska.rokoska@post.cz) <mailto:kokoska.rokoska@post.cz (kokoska.rokoska@post.cz)>> wrote:




seven napsal(a):
Quote:
You are not alone, I vote 1.

And there's a similer variable in conference:

<!--Can be | delim of waste|mute|deaf waste will always
transmit data to each channel
even during silence -->
<!--<param name="member-flags" value="waste"/>-->

Thank you very much, seven, for your support Smile

Best regards,

kokoska.rokoska


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AIM: anthm
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