forward from 1.0.4 we can't do debugging very easily.
I don't know all of the details of what you are trying to do but you are
hitting some race conditions because of the async nature of the socket
connection and the way you are using it.
On Wed, Dec 2, 2009 at 1:08 PM, Artem Shiyanov shiya
1 - config
2 - I've done this with programming
3 - suppose programming would be needed
Here is a bunch of code, search there ''barge
Artem
On Wed, Dec 2, 2009 at 11:34 AM, Nikolay Kondratyev k...@nstel.ru wrote:
Hi all,
I’m evaluating FS for our organization.
I must fulfill the
both types of variables are mutable
On Sun, Nov 22, 2009 at 2:25 PM, Lon Baker l...@kickasspixels.com wrote:
Are either global or regular channel variable mutable during a call?
Or can they only be set before and after?
Any clarification would help, since the existing wiki doesn't make it
I had almost the same problem- it was needed to record everything, even if
the record path doesn't exist - it was requested to create the needed path.
For this purpose I've used event_socket command api system ..., precisely,
api system mkdir -p path
And after this command I've started recording.
Hi there!
I've got annoying FS behavior:
There are 2 channels executing the same Java application (application itself
is an IVR). If I try to bridge them with uuid_bridged then both channels are
killed. Here is a log from FS console:
uuid_bridge 68587a9d-1d20-48f1-bdfc-72a2c027e1d2
this in a console log for us
and add an exact description of what you are doing in detail.
On Thu, Nov 5, 2009 at 11:44 AM, Artem Shiyanov shiya...@gmail.comwrote:
Hello!
I have to deal with classic problem: Leaking stream handle in FS
console. I also know the reason - firstly channel is sent
Here is rather big and, let's say, complete example of mod_java usage:
https://starpound.svn.sourceforge.net/svnroot/starpound/trunk/src/fs2agi
The goal of this project is to be a proxy between FreeSwitch and server
application which knows Asterisk AGI.
On Mon, Nov 2, 2009 at 2:53 PM,
Hello!
I've got strange problem:
In my app which talks to FreeSwitch via mod_socket there is such logic:
pre conditions: channel1 - newly created, parked channell - with help of
'uuid_create' and 'originate ... park'; channel2 is talking with channel3
for (channel in {array of channel1, channel2,
Hi there!
Please, suggest how to specify custom caller sip domain (logical) in
originate command.
I've been trying several alternatives but no one worked:
1) specify full sip address in
origination_caller_id_number=1...@uat.pbx.mblagov.starpoundtech.net - FS
adds its IP address so the
result From
Tested- it works!
Thanks a lot!!
On Mon, Oct 26, 2009 at 6:32 PM, mayamatakeshi mayamatake...@gmail.comwrote:
On Tue, Oct 27, 2009 at 12:20 AM, Artem Shiyanov shiya...@gmail.comwrote:
Hi there!
Please, suggest how to specify custom caller sip domain (logical) in
originate command
Finally!!
Thank you Michael, I didn't know about status app. It satisfies all my
desires.
And again,
thanks for all the community for the strong support!
Artem
On Tue, Oct 13, 2009 at 10:48 PM, Michael Collins m...@freeswitch.orgwrote:
On Tue, Oct 13, 2009 at 8:32 AM, Artem Shiyanov
the frequency was answered above.
On Mon, Oct 12, 2009 at 12:38 AM, Artem Shiyanov shiya...@gmail.comwrote:
Michael, Diego,
thanks for the rapid answers!
As far as I understand, enable_heartbeat app is launching
SESSION_HEARTBEAT events that will stop when the call will be cleared. Also
I heard
Hi all!
As it stays in wiki:
...
HEARTBEAT
Status information for freeswitch trigerred by freeswitch's heartbeat every
20 seconds.
...
Is there any way to customize timeout of HEARTBEAT events?
Thanks in advance,
Artem
___
FreeSWITCH-users mailing
Michael, Diego,
thanks for the rapid answers!
As far as I understand, enable_heartbeat app is launching
SESSION_HEARTBEAT events that will stop when the call will be cleared. Also
I heard that enable_heartbeat works only for calls with proxied media.
What I want is to monitor FreeSwitch status:
...@gmail.com wrote:
yes call the app as three_way like i said in the other thread.
On Tue, Sep 15, 2009 at 9:22 AM, Artem Shiyanov shiya...@gmail.comwrote:
Hello!
I'm trying to implement barge in functionality (see
http://www.yourdictionary.com/telecom/barge-in) with eavesdrop but
still
Hello!
I'm trying to implement barge in functionality (see
http://www.yourdictionary.com/telecom/barge-in) with eavesdrop but still
with no success.
The situation is:
- Person A calls to the extension:
extension name=some_ext
condition field=destination_number expression=^900.$
action
implement a media bug that replaces all the audio of
the channel by silence, but that'd require some C coding.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 19-Aug-09, at 7:13 AM, Artem Shiyanov wrote:
The point
in the conference.conf.xml to implement
your own features something like mute or kick. Or do you want to mute mute
other conference members like a moderator can do this.
BR
- Ursprüngliche Mail -
Von: Artem Shiyanov shiya...@gmail.com
An: freeswitch-users@lists.freeswitch.org
Hello all!
I'm trying to implement mute feature with mod_event_socket: I want
programmatically mute/unmute some channel in a call.. And I don't see any
other ways except to use conference room with special rule mute.
Anybody knows the better way?
Thanks,
Artem
, but this is
unwanted workaround for me..
So I wonder: is there any other (preferably through the SIP) way to mute
given SIP channel with Freeswitch?
Thanks for all,
Artem Shiyanov
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
:03 PM, Brian West br...@freeswitch.org wrote:
Can you try svn trunk... I know the remote FreeSWITCH isn't on trunk
also... due to the lines below.
/b
On Jun 26, 2009, at 12:59 PM, Artem Shiyanov wrote:
o=FreeSWITCH 3898736844884231 105343281763004076 IN IP4
192.168.147.130
s
Update again:
FS debug logs of the problematic part
http://pastebin.freeswitch.org/pastebin.php?dl=9512
Artem
On Mon, Jun 29, 2009 at 7:07 PM, Artem Shiyanov shiya...@gmail.com wrote:
I've tried with the snapshot (06.26.2009) - and situation had become even
worse - now both agents hear
Hello!
I got a problem with one way audio, symptoms are:
firstly play audio file to channel A (A is hears sound)
secondly bridge channel B with A (A doesn't hear B).
Environment:
- no NAT
- User Agents being used X-Lite, EyeBeam, SJphone - same result for all of
them- no audio, Wireshark shows
RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Artem
On Fri, Jun 26, 2009 at 9:25 PM, Artem Shiyanov shiya...@gmail.com wrote:
Hello!
I got a problem with one way audio, symptoms are:
firstly play audio file to channel A (A is hears sound)
secondly bridge channel B
Hi everyone!
In my environment I use FreeSwitch as media server and session border
controller. SIP routing is mostly done with my private B2BUA. The problem
itself is in my hold functionality. In details: A is calling to B:
!--
if the calling party is the called party, go to their VM
HOLD since we do not support it.
/b
On May 27, 2009, at 7:38 AM, Artem Shiyanov wrote:
Hi everyone!
In my environment I use FreeSwitch as media server and session border
controller. SIP routing is mostly done with my private B2BUA. The problem
itself is in my hold functionality. In details
...@freeswitch.org wrote:
Try not using RFC2543 HOLD since we do not support it.
/b
On May 27, 2009, at 7:38 AM, Artem Shiyanov wrote:
Hi everyone!
In my environment I use FreeSwitch as media server and session border
controller. SIP routing is mostly done with my private B2BUA. The problem
itself
won't hear anything if you press HOLD... the other caller you were
talking to will hear music.
/b
On May 27, 2009, at 11:45 AM, Artem Shiyanov wrote:
Probably mentioned sopftphones simply do not play incomming media when the
call is holded?
In general, is how should I hold a call using FS
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