Because switch_time_now() is in microseconds.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 14-Aug-09, at 11:16 PM, Michael Jerris wrote:
task->runtime = switch_epoch_time_now(NULL) + 10;
On Aug 14, 2009, at 10:19 PM,
Hi,
Can you provide us with a backtrace of the crash? you can open a bug
report on http://jira.freeswitch.org/
Also, if you want to play a prompt file and wait for a dtmf to accept
the call, there are variables called group_confirm_file and
group_confirm_key that will make the core take c
Please open a bug on jira.freeswitch.org
mike
On Aug 14, 2009, at 6:27 PM, Marc Orenberg wrote:
Hello,
I'm trying to play a prompt to the B-leg of a bridged call in Python.
I place the call to the B-leg, play the prompt, and then bridge it
with the A-Leg, but then FreeSWITCH crashes when t
task->runtime = switch_epoch_time_now(NULL) + 10;
On Aug 14, 2009, at 10:19 PM, mark morreny wrote:
Hi Michael,
The following code was executed once, but not after the next 10 s.
SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starti
Hi Michael,
The following code was executed once, but not after the next 10 s.
SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starting to flush
data buffer...\n");
task->runtime = switch_time_now() + 10;
}
Any suggestion why?
Th
Hello,
I'm trying to play a prompt to the B-leg of a bridged call in Python.
I place the call to the B-leg, play the prompt, and then bridge it with the
A-Leg, but then FreeSWITCH crashes when the call is completed.
Here's the code I'm using:
def
bridge_call_with_prompt(session,carrier,call
svn 14521: skypiax: compiles on windoz, not yet tested (on windoz)
On Fri, Aug 14, 2009 at 7:43 PM, Giovanni Maruzzelli
wrote:
> Hi FreeSWITCHers,
>
> all the users of mod_skypiax are kindly requested to test the svn trunk 14519.
>
> It contains a lot of changes meant to add stability and robus
All right, I'm confused.
The RTP timeout parameters have no documentation, and from the names, I'd
have guessed that they hang up after a specified amount of time, not send
some other signal.
The session-timeout timer talks about calls expiring, and sending another
SIP invite, which I don't think
That sounds horrible. There are settings both in sip/rtp and in
conference to do this already.
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#session-timeout
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Fi
Hi Rupa,
What about my suggestion above introduce a "api_after_bridge" event
that fires when the switch_ivr_uuid_bridge() bridges to the two sofia
channels that Mathieu mentioned?
Is that suggestion just way off the mark? If possible that would allow
me to move forward - although I agree that sup
Hi there,
Has anyone found a good solution for processing recordings/voicemail with
automatic speech recognition software?
I am researching numerous solutions, both open source and commercial, but
was wondering if I missed anything obvious.
Lon Baker
Kickass Pixels
-
(office) +1-415-287-0973
(mob
Thanks, Brian. If you don't mind my asking, what is the ultimate purpose
of local_ip_v4, then?
Brian West wrote:
Don't use local_ip_v4 then... please hard code the param in the sofia
profile because the value of local_ip_v4 can change.
/b
On Aug 14, 2009, at 7:14 AM, Carlos S. Antunes wrote
I didn't see any SIP session timers in the wiki. Since I'm already using the
event socket for control, my current plan is to use sched_api to play a file
with a short (20ms?) clip of silence, capture the play_file event and use it
to reschule another one for a couple of seconds later.
I'll let you
On Fri, Aug 14, 2009 at 11:20 AM, Tina Martinez wrote:
> Michael,
>
> Thanks again for bearing with my novice perspective on this.
>
> I was able to achieve the link between two FS servers as intended.
> However, I
> was not able to setup a "new" dialplan file as you described. I had to
> place
How exactly are you setting the var? It should be set on the b-leg,
such as using [sip_contact_user=xxx] on the originate line.
Mike
On Aug 14, 2009, at 12:02 PM, Juan Backson wrote:
> Hi,
>
> I would like to set outbound INVITE with customer Contact: field. I
> did sip_contact_user="20333
thats in seconds.
Mike
On Aug 14, 2009, at 8:32 AM, mark morreny wrote:
Hi,
Thank you for your help.
I get that too, but the callback does not execute the second time.
When I do task->runtime = switch_time_now() + 10;, what does +10
mean? Does it mean 10 s or 10 mins?
Thanks,
Mark
On
It would appear that different versions of opensolaris / compiler /
32/64 bit handle this totally differently. I have tested this both 32
and 64 on sun studio when I made the change and it works, seems gcc
wants different format specifiers. The patch that caused this is :
http://fisheye.fr
Issue is we don't handle progress and progress media differently,
maybe we should. The message is however harmless, but annoying and
should probably be dealt with a bit better. Patches welcome.
Mike
On Aug 13, 2009, at 9:04 PM, Moises Silva wrote:
Yes, agreed, but there is no point in sen
I know sangoma is working on their new pri stack, maybe it has proper
qsig support. The openzap isdn stack does not right now and I was
under the impression that the libpri support was pretty limited, but
have no direct knowledge there.
Mike
On Aug 13, 2009, at 3:34 PM, Ryan Wagoner wrote:
Announcing the release of FreeSWITCH Console in the Apple Application
Store. The application is FREE and allows you to connect to a
FreeSWITCH event socket layer module that is bound to an external
interface. Great for development purposes and general remote debugging.
Blog announcement:
http
Hi FreeSWITCHers,
all the users of mod_skypiax are kindly requested to test the svn trunk 14519.
It contains a lot of changes meant to add stability and robustness,
toward a production environment.
Let me know how your feelings, and please add to the Jira any possible
bug/issue/etc.
Thanks to y
Michael,
Thanks again for bearing with my novice perspective on this.
I was able to achieve the link between two FS servers as intended. However, I
was not able to setup a "new" dialplan file as you described. I had to place
the
script into the default.xml dialplan to get it to work. Is there
Hi,
I would like to set outbound INVITE with customer Contact: field. I did
sip_contact_user="204545", but it still did not work. I am using 1.0.4
release.
Is this a known issue?
br,
JB
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.fre
No I'm going to have to put them on torrent. Too many people were
watching them off of files-sync on their media center PC's and chewing
bandwidth galore.
/b
On Aug 14, 2009, at 8:37 AM, William Suffill wrote:
> Given that people were downloading directly from the source that feeds
> the CD
Don't use local_ip_v4 then... please hard code the param in the sofia
profile because the value of local_ip_v4 can change.
/b
On Aug 14, 2009, at 7:14 AM, Carlos S. Antunes wrote:
> Hi Brian!
>
> Thank you for your quick response. I ended up defining "local_ip_v4"
> at the top of vars.xml an
Given that people were downloading directly from the source that feeds
the CDN (files.freeswitch.org is hosted by a CDN) it will be some
time before the files appear available again. It became a problem
with how much bandwidth was being used without using the CDN. Best to
just wait I know it's be
Got this:
Forbidden
You don't have permission to access /cluecon_2009/presentations/
Dale_Building_FreeSWITCH_App_Lua.pptx on this server.
Apache/2.2.3 (CentOS) Server at files-sync.freeswitch.org Port 80
$ ping files.freeswitch.org
PING filessync.freeswitch.netdna-cdn.com (69.174.57.101): 56 d
Hi,
Thank you for your help.
I get that too, but the callback does not execute the second time.
When I do task->runtime = switch_time_now() + 10;, what does +10 mean? Does
it mean 10 s or 10 mins?
Thanks,
Mark
On Wed, Aug 12, 2009 at 11:09 PM, Mathieu Rene wrote:
> Hi,
> I did the same thin
Hi Brian!
Thank you for your quick response. I ended up defining "local_ip_v4" at
the top of vars.xml and set "bind_server_ip" to "local_ip_v4" as well.
Is this the best way to go about selecting an IP address for FS to bind to?
In any case, even though it doesn't appear to affect operation,
Hi All,
Anyone have any ideas about this?
Thanks
Bruce
vmorales wrote:
> Hi Bruce,
>
> I am having similar issues trying build freeswitch 1.0.4 on Solaris
> x86 as well. I sent some information over the mailing list, and I
> received a response from Michal Bielicki (attached), stating he'd tes
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