This bug has been now closed out in jira due to no response for requested
information. If you wish to resolve this issue please follow up on your bugs
when information is requested.
Mike
On Oct 12, 2009, at 12:04 PM, Maciej Aniserowicz wrote:
>
> Nope, I wanted to make sure that this is inde
Nope, I wanted to make sure that this is indeed a bug. I opened an issue in
JIRA before regarding some other matter and it turned out to be my mistake,
so I decided to try mailing list first this time.
MA
Brian West wrote:
>
> Did you open a jira and attach all the info?
>
> /b
>
> On Oct 12
Did you open a jira and attach all the info?
/b
On Oct 12, 2009, at 3:47 AM, Maciej Aniserowicz wrote:
Yes, I confirmed that with Wireshark (filter "rtp and ip.src ==
). RTP packets are sent every 20ms.
MAniserowicz
___
FreeSWITCH-users mailing
Yes, I confirmed that with Wireshark (filter "rtp and ip.src == ).
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] Bad
can you confirm from an rtp packet trace that they are all really
sending 20ms?
Mike
On Oct 10, 2009, at 6:04 AM, Maciej Aniserowicz wrote:
>
> Hi,
> Here are the messages with a:ptime parameter. All the calls are
> started by
> commands sent through socket.
> I'm not sure if this is all inf
Hi,
Here are the messages with a:ptime parameter. All the calls are started by
commands sent through socket.
I'm not sure if this is all information you need, please let me know if
something is missing here and I'll post that.
1) starting connection with x-lite (number 2003, the eavesdropper):
you probably have some device lying about ptime everywhere
look at a sip trace an pay especially close attention to ptime:x param in
sdp
if you don't understand this just attach it here
execute the following at the cli
sofia profile internal siptrace on
sofila loglevel debug
On Thu, Oct 8, 2009
It's the same on the trunk (the last rev I used was not so old anyway).
Codecs are the same on both legs:
read codec/read rate: PCMU 8000
write codec/write rate: PCMU8000
MA
Michael Jerris wrote:
>
> What codecs are all the call legs using, also, please try current svn
> trun
What codecs are all the call 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 e
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 being eavesdropped
I did not change any codec configuration, I just
That's is a somewhat vague position.
You did not mention which version of FreeSWITCH you are running, the phones
being used in your example, your configuration, the codecs in use etc.
BTW,
I think you should only ask one question at a time on this list. The list
is run by volunteers and it's sor
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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