On line 424 I think it needs to be changed from
if (!vval || !switch_true(vval)) {
to
if (!vval || switch_true(vval)) {
Other wise it works, thanks!
----- Original Message -----
From: "Anthony Minessale" <[email protected]>
To: [email protected]
Sent: Friday, February 6, 2009 1:48:31 PM GMT -07:00 US/Canada Mountain
Subject: Re: [Freeswitch-users] mod_shout delay in trunk
Thanks,
We appreciate the positive feedback!
if you revert the change I suggested and update i added a new variable
enable_file_write_buffering=false
set this variable on the channel before you start recording it with the set
application or in the dialstring in {}
on outbound calls and it should skip the buffering.
Could you test it for me and confirm it works?
Thank you
On Fri, Feb 6, 2009 at 2:36 PM, < [email protected] > wrote:
That worked great!
I wanted to say just how awesome Freeswitch is, I have been doing voip related
development with SIP since 2000 and this is by far the most well designed piece
of voip software I have used or developed on. I currently have a homegrown sip
server built on the NIST sip stack with Sun's JMF libraries for RTP processing.
95% of the code and complexity is handling the SIP and RTP sessions, the other
5% is the final application logic and what is most important to me. By letting
freeswitch do whats its good at (call routing, sip and media handling) it
allows me to focus on what I'm good at (what should we do with those streams,
like record them). I have been bragging about this project to anybody who will
listen!
Dan-
----- Original Message -----
From: "Anthony Minessale" < [email protected] >
To: [email protected]
Sent: Friday, February 6, 2009 1:07:44 PM GMT -07:00 US/Canada Mountain
Subject: Re: [Freeswitch-users] mod_shout delay in trunk
edit switch_ivr_play_say.c line 423
comment the line out and recompile.
Tell me if it helps you and i will consider making it configurable.
On Fri, Feb 6, 2009 at 2:01 PM, < [email protected] > wrote:
For me it is. For what I'm using it for I can tolerate around a second or two
delay. I have the icecast server setup to only buffer 1K for their on-connect
burst as well as my flash/flex player to only buffer 1k (yes I might as well
not buffer at all, which I may end up doing). In 1.0.2 this worked very well.
Is this buffer configurable? If not, where is it being set?
Thanks
Dan-
----- Original Message -----
From: "Brian West" < [email protected] >
To: [email protected]
Sent: Friday, February 6, 2009 12:47:53 PM GMT -07:00 US/Canada Mountain
Subject: Re: [Freeswitch-users] mod_shout delay in trunk
Let me clarify.. yes this is normal file buffering was added so we wouldn't
thrash your hard drive with tiny bits of data when recording calls so now it
buffers and writes larger chunks to disk. This is why you have this delay which
is 100% normal.... is realtime a critical thing? It is shout cast so you know
it doesn't have to be realtime.. in fact some clients will buffer a little bit
anyway and add to it.
/b
On Feb 6, 2009, at 1:43 PM, [email protected] wrote:
I have, do you know what would have changed between 1.0.2 and trunk that would
cause the buffer to change? Also if its not in mod_shout.c (which I copied from
1.0.2 to trunk for testing with no luck), where else would fs be buffering? One
thing I have noticed is that in 1.0.2 as soon as the dial plan hits my record
statement I see mod_shout logging that it has connected to the icecast server,
in trunk it takes about 5 seconds to see the same log mesage. Below is my
current svn info
Path: .
_______________________________________________ Freeswitch-users mailing list
[email protected]
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
[email protected]
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:[email protected]
GTALK/JABBER/ PAYPAL:[email protected]
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:[email protected]
iax:[email protected]/888
googletalk:[email protected]
pstn:213-799-1400
_______________________________________________ Freeswitch-users mailing list
[email protected]
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
[email protected]
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:[email protected]
GTALK/JABBER/ PAYPAL:[email protected]
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:[email protected]
iax:[email protected]/888
googletalk:[email protected]
pstn:213-799-1400
_______________________________________________ Freeswitch-users mailing list
[email protected]
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
[email protected]
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
