config seems to
work. I thought that the shortest configured value should cause the timeout,
but it's not the case. Am I missing something, or is this the correct
behavior?
Regards,
Maciej Aniserowicz
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to
work like 2nd user
All of them are simulated by dialplan extensions (using answer and playback
tools), but the same thing happends for xlite or cisco phone.
Maciej Aniserowicz
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Sent
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
Correct - compiled but did not run. Works fine now.
I'll see if the error shows up again and let you know if it does.
Thanks,
MA
Anthony Minessale wrote:
won't compile or won't run?
maybe you should try rebuilding it.
On Tue, Oct 27, 2009 at 9:55 AM, Maciej Aniserowicz
]
To: Maciej Aniserowicz
Sent: Monday, October 26, 2009 10:32 PM
Subject: Re: [Freeswitch-users] Can not record session. Media not enabled on
channel.
On Mon, Oct 26, 2009 at 12:21 AM, Maciej Aniserowicz [hidden email] wrote:
Yes, I can confirm - this exact error occurs each
to be ready?
mercutioviz wrote:
On Fri, Oct 23, 2009 at 12:36 AM, Maciej Aniserowicz
maciej.aniserow...@gmail.com wrote:
The dialplan is very simple:
extension name=Recording test
condition field=destination_number
expression=^11\d
and post it in pastebin.
-MC
On Tue, Oct 20, 2009 at 3:30 AM, Maciej Aniserowicz
maciej.aniserow...@gmail.com wrote:
Hello,
I am using the same set of extensions for testing the system during
development, they include XLite, Cisco sip phone and several extensions
that
just play some
.
Like I wrote before, I don't change any codec settings and always use the
same set of devices/emulators.
What can cause this message?
Thanks,
Maciej Aniserowicz
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Sent from
Yes, I confirmed that with Wireshark (filter rtp and ip.src == device ip).
RTP packets are sent every 20ms.
MAniserowicz
- Original Message -
From: Michael Jerris (via Nabble)
To: Maciej Aniserowicz
Sent: Monday, October 12, 2009 12:21 AM
Subject: Re: [Freeswitch-users
, 2009, at 3:47 AM, Maciej Aniserowicz wrote:
Yes, I confirmed that with Wireshark (filter rtp and ip.src ==
device ip). RTP packets are sent every 20ms.
MAniserowicz
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http
legs using, also, please try current svn
trunk.
Mike
On Oct 7, 2009, at 3:39 AM, Maciej Aniserowicz wrote:
Sorry about posting several questions at once, I wasn't aware it's
rude.
Let's concentrate on this issue then.
I use FS rev 14994. Phones on extensions:
1) x-lite
2
of you for your answers.
MA
- Original Message -
From: mercutioviz (via Nabble)
To: Maciej Aniserowicz
Sent: Thursday, October 08, 2009 7:06 PM
Subject: Re: [Freeswitch-users] gateway FS informs it's client FS about users
hanged up with a long delay
On Thu, Oct 8
Both of the instances are run on the same machine, i just changed the default
ports they use. Can anything else cause this strange behavior?
MA
Michael Jerris wrote:
Incorrect NAT configuration so one of the boxes is not actually
getting a BYE.
On Oct 5, 2009, at 3:13 AM, Maciej
Yes, I know that FS deletes short files. I just don't know why the file is so
small... it is always 388 bytes, no matter how long the session lasts.
MA
Michael Jerris wrote:
switch_ivr_async.c:480
On Oct 5, 2009, at 3:16 AM, Maciej Aniserowicz wrote:
Hi,
When I record a call in FS
.
Mike
On Oct 7, 2009, at 3:39 AM, Maciej Aniserowicz wrote:
Sorry about posting several questions at once, I wasn't aware it's
rude.
Let's concentrate on this issue then.
I use FS rev 14994. Phones on extensions:
1) x-lite
2) cisco sip phone
3) audio played by fs to the extension
use the standard one that
comes with both FS and the phones.
Some time ago someone on FS irc channel told me that this is just how FS
eavesdropping works... from your response I understand that this is not
entirely true?
Maciej Aniserowicz
Anthony Minessale wrote:
That's is a somewhat vague
Hi,
When I record a call in FS, it only creates a 388-byte-long wav file. The
conversation is no written there, and FS deletes the file when the session
finishes.
What can cause this strange behavior?
Br/
Maciej Aniserowicz___
FreeSWITCH-users mailing
Hi,
When I use two FreeSWITCH instances ('internal' and 'external'), all users
register to the 'external' instance which acts as a gateway by 'internal'
instance (which in turn is controlled by my applicaiton with commands sent by
socket).
When user hangs up, the 'hanged up' event is propagated
Hello,
When I use eavesdropping in FreeSWITCH, the sound quality is really bad. Is
there any way to improve it? Is this a known problem?
Br/
Maciej Aniserowicz___
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