Hi all,
The problem is solved. I ask the softswitch to send only sdp in INVITE
message, then It works.
I think sofia doesn't support multipart content currently. is it right?
2009/11/2 Lei Tang lei.tl...@gmail.com
Hi Daniel.
Sure. pls email me to tl...@hotmail.com.
2009/11/2 Zeng, Qinglan
No, I'm Seven and never used hangup hook. you must had though I was Dome.
Sorry, I'm not tend to hijack this thread, just though it's the same topic.
2009/11/2 Anthony Minessale anthony.miness...@gmail.com
We already concluded its your unacceptabe use of originate in hangup hook
right?
On
Hi
pastebin:
http://pastebin.freeswitch.org/10926
and
http://pastebin.freeswitch.org/10927
.We invoke calls from one voip phone to cell phone, and vice versa, but
when i make inbound and outbound connection in nearly same time
something goes wrong with chanells
Thanks
Michael Collins
Hi,
Can you please paste me your sample java dialplan code that work for you ?
..coz m also facing the same problem.
My mod_java is loaded properly.
Also /usr/lib/jvm/java-1.5.0-gcj-4.3-1.5.0.0/jre/lib/i386/client/libjvm.so and
freeswitch.jar in java.conf.xml is specified properly.
I have
Hi,
i've done more and more tests ... the result is the same :\
I've tried previous freeswitch version (1.0.2, 1.0.3), lastest stable
(1.0.4) and with svn (updated at revision 15315 while openzap revision
is 847).
I've tried with ubuntu zaptel modules (1.4.10), with and without octvqe
soft
Thanks for you answers guys,
I test the parameters you suggested
but still no audio due to the lack of reINVITE. By the way I'm using
1.0.4 but I also tried 1.0.5pre3.
One particular condition is that there is no on-hold before the Blind Transfer.
Regards,
Humberto
param name=media-option
That is correct.
Mike
On Nov 2, 2009, at 4:24 AM, Lei Tang wrote:
Hi all,
The problem is solved. I ask the softswitch to send only sdp in
INVITE message, then It works.
I think sofia doesn't support multipart content currently. is it
right?
Please re-try with latest svn trunk.
Mike
On Nov 2, 2009, at 9:36 AM, Humberto Quintana wrote:
Thanks for you answers guys,
I test the parameters you suggested
but still no audio due to the lack of reINVITE. By the way I'm using
1.0.4 but I also tried 1.0.5pre3.
One particular
Every time you have stuck channels at the last state it means something took
control
of the thread and did not release it.
revisions other that current SVN trunk are not possible to debug because
over one thousand changes have occurred since then.
On Mon, Nov 2, 2009 at 4:03 AM, Seven Du
Yes, I think I did. However here is what furthur testing revelas. If I dial in
from ATT cell phone, I do not see any DTMF using Don's IVR.xml.conf to call my
conf app. But when I dial the same number using a Verizon Cell, it works.
When I dial a number that is provisioned to call the Conf App
you know I have heard this before... It seems to ONLY be ATT
/b
On Nov 2, 2009, at 9:54 AM, Ujjval Karihaloo wrote:
Yes, I think I did. However here is what furthur testing revelas. If
I dial in from ATT cell phone, I do not see any DTMF using Don's
IVR.xml.conf to call my conf app. But
Any suggestion from anyone please?
Thank you,
Dorn B
- Original Message
From: DJB djbin...@yahoo.com
To: freeswitch-users@lists.freeswitch.org
Sent: Fri, October 30, 2009 11:31:12 AM
Subject: Re: [Freeswitch-users] Freeswitch in signaling path only
Now i have as follows, but it's
Anthony,
Yes, if you can advise, how would I detect whether it's going out to
192.168.1.4 or 192.168.1.5 without having to activate b-leg of the CDRs.
Thank you,
Dorn B.
From: Anthony Minessale anthony.miness...@gmail.com
To:
if you enable debug on the cdr_csv module you will get a big dump of all the
data you have available and you may be able to pick something out that
indicates which one it was.
On Mon, Nov 2, 2009 at 10:52 AM, DJB djbin...@yahoo.com wrote:
Anthony,
Yes, if you can advise, how would I detect
I am really sorry. I did not mean to rush or anything. I've had a problem
with my email many times, so I just want to make sure that my email gets there.
Regards,
Dorn B.
From: Anthony Minessale anthony.miness...@gmail.com
To:
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,
Is starpound involved in the FS Community?
/b
On Nov 2, 2009, at 12:51 PM, Artem Shiyanov wrote:
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
Rob:
Once I have the Moderator and Participants logged on, how do I invoke the
moderator previlidges, LIk esay muting everyone/someone or kicking someone out
of the Conf and the like?
Ujjval Karihaloo
VP Voice Engineering
IP Phone: +13032428610
E-Fax: +17202391690
SimpleSignal Inc.
88
That was it. My sip provider applied the patch to his Asterisk server
that was referenced in the link you was so kind to provide, and again
everything worked as it should.
Thank you very much!
Ivan
Den 1. nov. 2009 kl. 21:20 skrev Anthony Minessale:
Session-Expires: -1;refresher=uas
On Mon, Nov 2, 2009 at 12:58 PM, Ivan C Myrvold i...@myrvold.org wrote:
That was it. My sip provider applied the patch to his Asterisk server that
was referenced in the link you was so kind to provide, and again everything
worked as it should.
Thank you very much!
This is why Tony's
Hi Mike,
I re-tried with trunk rev 15319 but I got almost the same behavior: There is
now a reINVITE (with FS' SDP) going to A when the REFER is accepted. But still
there is no reINVITE for A (with C's SDP) after the call from FS to C is
established.
Anyway, we decided for now to do a
please try r15326
I think i have it working.
I recommend for optimal results you set bypass_media_after_bridge=true
either as a global or in your DP in place of bypass_media=true
On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana hjqlo...@hotmail.comwrote:
Hi Mike,
I re-tried with trunk
Hi,
Unfortunately getting the newest version did not solve the problem: Can not
record session. Media not enabled on channel. error still appears
sometimes.
MA
Maciej Aniserowicz wrote:
Correct - compiled but did not run. Works fine now.
I'll see if the error shows up again and let you
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