Re: [Freeswitch-users] no RTP send during Voice Mail recording
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:
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.roko...@post.cz
wrote:
Thank you very much, Dave, for your help!
Mentioned modification did the trick and all works as I wish :-)
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 :-)
Is it normal or should I investigate what is wrong?
Best regards,
kokoska.rokoska
David Knell napsal(a):
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
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 :-)
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 :-) know what I'm doing wrong?
Thanks once more, dujinfang, for your help!
Best regards,
kokoska.rokoska
dujinfang napsal(a):
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:
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):
sigh,
see r13144
On Fri, Apr 24, 2009 at 3:43 AM, kokoska rokoska
kokoska.roko...@post.cz mailto:kokoska.roko...@post.cz
wrote:
seven napsal(a):
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 :-)
Best regards,
kokoska.rokoska
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ClueCon http://www.cluecon.com/
AIM: anthm
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Re: [Freeswitch-users] no RTP send during Voice Mail recording
ok done
set the var to
true / false / desired silence factor
On Sun, Apr 26, 2009 at 7:09 AM, dujinfang dujinf...@gmail.com wrote:
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:
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.roko...@post.czwrote:
Thank you very much, Dave, for your help!
Mentioned modification did the trick and all works as I wish :-) 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 :-)
Is it normal or should I investigate what is wrong?
Best regards,
kokoska.rokoska
David Knell napsal(a):
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
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
:-)
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 :-) know what I'm doing wrong?
Thanks once more, dujinfang, for your help!
Best regards,
kokoska.rokoska
dujinfang napsal(a):
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:
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):
sigh,
see r13144
On Fri, Apr 24, 2009 at 3:43 AM, kokoska rokoska
kokoska.roko...@post.cz mailto:kokoska.roko...@post.cz wrote:
seven napsal(a):
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 :-)
Best regards,
kokoska.rokoska
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FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
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Re: [Freeswitch-users] no RTP send during Voice Mail recording
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:
ok done
set the var to
true / false / desired silence factor
On Sun, Apr 26, 2009 at 7:09 AM, dujinfang dujinf...@gmail.com
wrote:
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:
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.roko...@post.cz
wrote:
Thank you very much, Dave, for your help!
Mentioned modification did the trick and all works as I wish :-)
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 :-)
Is it normal or should I investigate what is wrong?
Best regards,
kokoska.rokoska
David Knell napsal(a):
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
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 :-)
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 :-) know what I'm doing wrong?
Thanks once more, dujinfang, for your help!
Best regards,
kokoska.rokoska
dujinfang napsal(a):
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:
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):
sigh,
see r13144
On Fri, Apr 24, 2009 at 3:43 AM, kokoska rokoska
kokoska.roko...@post.cz mailto:kokoska.roko...@post.cz
wrote:
seven napsal(a):
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 :-)
Best regards,
kokoska.rokoska
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FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
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FreeSWITCH Developer Conference
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http://iax:gu...@conference.freeswitch.org/888
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Re: [Freeswitch-users] no RTP send during Voice Mail recording
sigh, see r13144 On Fri, Apr 24, 2009 at 3:43 AM, kokoska rokoska kokoska.roko...@post.czwrote: seven napsal(a): 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 :-) Best regards, kokoska.rokoska ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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
Re: [Freeswitch-users] no RTP send during Voice Mail recording
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): sigh, see r13144 On Fri, Apr 24, 2009 at 3:43 AM, kokoska rokoska kokoska.roko...@post.cz mailto:kokoska.roko...@post.cz wrote: seven napsal(a): 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 :-) Best regards, kokoska.rokoska ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto: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_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net http://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 http://iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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
Re: [Freeswitch-users] no RTP send during Voice Mail recording
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: 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): sigh, see r13144 On Fri, Apr 24, 2009 at 3:43 AM, kokoska rokoska kokoska.roko...@post.cz mailto:kokoska.roko...@post.cz wrote: seven napsal(a): 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 :-) Best regards, kokoska.rokoska ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto: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_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net http://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 http://iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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
Re: [Freeswitch-users] no RTP send during Voice Mail recording
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 :-)
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 :-) know what I'm doing wrong?
Thanks once more, dujinfang, for your help!
Best regards,
kokoska.rokoska
dujinfang napsal(a):
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:
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):
sigh,
see r13144
On Fri, Apr 24, 2009 at 3:43 AM, kokoska rokoska
kokoska.roko...@post.cz mailto:kokoska.roko...@post.cz wrote:
seven napsal(a):
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 :-)
Best regards,
kokoska.rokoska
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FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_miness...@hotmail.com
mailto:msn%3aanthony_miness...@hotmail.com
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FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org
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Re: [Freeswitch-users] no RTP send during Voice Mail recording
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
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 :-)
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 :-) know what I'm doing wrong?
Thanks once more, dujinfang, for your help!
Best regards,
kokoska.rokoska
dujinfang napsal(a):
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:
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):
sigh,
see r13144
On Fri, Apr 24, 2009 at 3:43 AM, kokoska rokoska
kokoska.roko...@post.cz mailto:kokoska.roko...@post.cz wrote:
seven napsal(a):
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 :-)
Best regards,
kokoska.rokoska
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Re: [Freeswitch-users] no RTP send during Voice Mail recording
Thank you very much, Dave, for your help!
Mentioned modification did the trick and all works as I wish :-) 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 :-)
Is it normal or should I investigate what is wrong?
Best regards,
kokoska.rokoska
David Knell napsal(a):
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
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 :-)
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 :-) know what I'm doing wrong?
Thanks once more, dujinfang, for your help!
Best regards,
kokoska.rokoska
dujinfang napsal(a):
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:
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):
sigh,
see r13144
On Fri, Apr 24, 2009 at 3:43 AM, kokoska rokoska
kokoska.roko...@post.cz mailto:kokoska.roko...@post.cz wrote:
seven napsal(a):
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 :-)
Best regards,
kokoska.rokoska
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AIM: anthm
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Re: [Freeswitch-users] no RTP send during Voice Mail recording
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.roko...@post.czwrote:
Thank you very much, Dave, for your help!
Mentioned modification did the trick and all works as I wish :-) 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 :-)
Is it normal or should I investigate what is wrong?
Best regards,
kokoska.rokoska
David Knell napsal(a):
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
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
:-)
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 :-) know what I'm doing wrong?
Thanks once more, dujinfang, for your help!
Best regards,
kokoska.rokoska
dujinfang napsal(a):
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:
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):
sigh,
see r13144
On Fri, Apr 24, 2009 at 3:43 AM, kokoska rokoska
kokoska.roko...@post.cz mailto:kokoska.roko...@post.cz wrote:
seven napsal(a):
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 :-)
Best regards,
kokoska.rokoska
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Re: [Freeswitch-users] no RTP send during Voice Mail recording
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/-- On Apr 24, 2009, at 1:40 PM, kokoska rokoska wrote: I'd like to ask: Are there any plans to implement such feature/ variable, or I'm the only one who needs it? Best regards, kokoska.rokoska ___ 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
Re: [Freeswitch-users] no RTP send during Voice Mail recording
seven napsal(a): 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 :-) Best regards, kokoska.rokoska ___ 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
Re: [Freeswitch-users] no RTP send during Voice Mail recording
Anthony Minessale napsal(a): it's nothing to do with vad, it's simply how FS works. It's a waste to encode and send zeros into the channel while it's recording. Also, It's unreasonable to have such a short timeout. I understand it's not your fault, I am just letting you know. It would be possible to add a patch to create a channel variable like NDLB_waste_bandwidth_while_recording or something but it does not exist today. I'd like to ask: Are there any plans to implement such feature/variable, or I'm the only one who needs it? Best regards, kokoska.rokoska ___ 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
Re: [Freeswitch-users] no RTP send during Voice Mail recording
kokoska.rokoska napsal(a): Hi all, I fall into trouble with voice mail. It looks like FreeSWITCH sends no RTP during Voice Mail recording and thus the calls from my TSPs are cut off in the middle of the recording due to lack of RTP activity (based on providers tolerancy it happens in 5 to 20 seconds). I tried to set VAD to none in all sofia profiles but it doesn't help. Are there any other settings I have to use to force FreeSWITCH to send RTP back (silence, CNG or what ever :-) during VM recording? BTW: I'm on current trunk. Hi all, until previous message I have tried all combinations of VAD settings and VM recording format and still no luck: using ngrep I can't see any RTP packetes going from FreeSWITCH during VM recording = calls are dropped by my TSPs after few seconds. Could you, please, point me to some other direction where should I experient? Or is it desired behaviour of FreeSWITCH? Best regards, kokoska.rokoska ___ 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
Re: [Freeswitch-users] no RTP send during Voice Mail recording
it's nothing to do with vad, it's simply how FS works. It's a waste to encode and send zeros into the channel while it's recording. Also, It's unreasonable to have such a short timeout. I understand it's not your fault, I am just letting you know. It would be possible to add a patch to create a channel variable like NDLB_waste_bandwidth_while_recording or something but it does not exist today. On Mon, Apr 20, 2009 at 11:03 AM, kokoska rokoska kokoska.roko...@post.czwrote: kokoska.rokoska napsal(a): Hi all, I fall into trouble with voice mail. It looks like FreeSWITCH sends no RTP during Voice Mail recording and thus the calls from my TSPs are cut off in the middle of the recording due to lack of RTP activity (based on providers tolerancy it happens in 5 to 20 seconds). I tried to set VAD to none in all sofia profiles but it doesn't help. Are there any other settings I have to use to force FreeSWITCH to send RTP back (silence, CNG or what ever :-) during VM recording? BTW: I'm on current trunk. Hi all, until previous message I have tried all combinations of VAD settings and VM recording format and still no luck: using ngrep I can't see any RTP packetes going from FreeSWITCH during VM recording = calls are dropped by my TSPs after few seconds. Could you, please, point me to some other direction where should I experient? Or is it desired behaviour of FreeSWITCH? Best regards, kokoska.rokoska ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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
Re: [Freeswitch-users] no RTP send during Voice Mail recording
Anthony Minessale napsal(a): it's nothing to do with vad, it's simply how FS works. Thank you very much, Anthony, for explanation! It's a waste to encode and send zeros into the channel while it's recording. Also, It's unreasonable to have such a short timeout. Yes, I understand. But can do nothing with it :-) I understand it's not your fault, I am just letting you know. Like I wrote - I should live with it. It would be possible to add a patch to create a channel variable like NDLB_waste_bandwidth_while_recording or something but it does not exist today. Interesting variable name :-) This will waste bandwidth, I'm sure, but will also save my life (from not so happy users). And from shame to go back to, I am ashamed to write it, * :-) Thanks once more, Anthony, for your help and useful informations! Best regards, kokoska.rokoska ___ 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
Re: [Freeswitch-users] no RTP send during Voice Mail recording
Hi kokoska Actually, you can request your VSP to set the rtptimeout or whatever parameter in their SIP server to a reasonable value such as 300 seconds as 5 minutes should be enough for most standard business voice mail service, otherwise you should wait for live calls instead of leaving voice messages. In * they have the following setting which is default to 60 seconds if nothing changed rtptimeout=300 ; Terminate call if 60 seconds of no RTP or RTCP activity ; on the audio channel ; when we're not on hold. This is to be able to hangup ; a call in the case of a phone disappearing from the net, ; like a powerloss or grandma tripping over a cable. This works with one of my ITSP as they provide SIP trunking via * Hope this helps. Chris On Mon, Apr 20, 2009 at 12:48 PM, kokoska rokoska kokoska.roko...@post.czwrote: Anthony Minessale napsal(a): it's nothing to do with vad, it's simply how FS works. Thank you very much, Anthony, for explanation! It's a waste to encode and send zeros into the channel while it's recording. Also, It's unreasonable to have such a short timeout. Yes, I understand. But can do nothing with it :-) I understand it's not your fault, I am just letting you know. Like I wrote - I should live with it. It would be possible to add a patch to create a channel variable like NDLB_waste_bandwidth_while_recording or something but it does not exist today. Interesting variable name :-) This will waste bandwidth, I'm sure, but will also save my life (from not so happy users). And from shame to go back to, I am ashamed to write it, * :-) Thanks once more, Anthony, for your help and useful informations! Best regards, kokoska.rokoska ___ 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
Re: [Freeswitch-users] no RTP send during Voice Mail recording
Thank you very much, Chris, for your reply! Chris Chen napsal(a): Hi kokoska Actually, you can request your VSP to set the rtptimeout or whatever parameter in their SIP server to a reasonable value such as 300 seconds as 5 minutes, I'm afraid (well, I'm pretty sure) non of them want to do it, because they need very accurate billing and this is simpliest way how to do it - kill calls without RTP i few seconds. should be enough for most standard business voice mail service, otherwise you should wait for live calls instead of leaving voice messages. In * they have the following setting which is default to 60 seconds if nothing changed rtptimeout=300 ; Terminate call if 60 seconds of no RTP or RTCP activity ; on the audio channel ; when we're not on hold. This is to be able to hangup ; a call in the case of a phone disappearing from the net, ; like a powerloss or grandma tripping over a cable. Yes, I know. I have spent some years with * in the past (from pre 1.0 release if I remember correctly :-). In my post I mean * ability to send faked audio during recording: transmit_silence_during_record=yes option in asterisk.conf This works with one of my ITSP as they provide SIP trunking via * None of my TSPs use Asterisk :-) Around me there are much more popular Cirpacks and Phonets - due to scalability, features, SS7 support etc... Hope this helps. Thanks once more, Chris, for your interest! Best regards, kokoska.rokoska ___ 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
