You can do all that. SRTP<>RTP doesnt depend on the codec since it
just decrypts the packet's contents and SIP/TLS is dependent on that
specific connection.
Math
On 20-Jun-09, at 12:42 AM, xbipin wrote:
>
> im just confused about certains tuff in freeswitch, firstly if we
> use TLS for
> S
im just confused about certains tuff in freeswitch, firstly if we use TLS for
SIP then FS can read such packets but if it were to act as gateway can it
communicate in plain SIP, so its like phone connected to FS using TLS and
then from FS to gateway in plain SIP?
secondly, if phone uses SRTP, can
Some of these things make sense is some scenarios but not others.
Most people are wanting to do full topology hiding, so we don't by
default pass very much across a bridge. I am interested in working on
this, feel free to contact me off list with your findings.
Mike
On Jun 17, 2009, at 11
Try using loopback endpoint for this test .
Mike
On Jun 17, 2009, at 10:00 AM, Tim B wrote:
Trying to do a local test for faxing. Keep getting an error.
default dialplan:
//inbound from remote box works fine
- connect asterisk box and fs box, then fax from asterisk to fs... OK
- a
You are right, it seams that it can not be done.
In the past I've tried to do something similiar but with no success.
Apperently the documentation is wrong.
If I have some time I will look at the code and I will give you some
feedback.
Cheers,
PS: The transcoding question has nothing to do
Ok so after many attempts of trial and error I narrowed it down to acls. So
when trying to orginate a call to the local FS extension it was getting blocked.
Adding the following allow with my freeswitch IP to the domains list allowed
the originate to take place.
acl.conf.xml:
Yes that is the extension defined in the default dialplan. It is setup like
explained under mod_fax... here is the actual definition:
http://pastebin.freeswitch.org/9450
> --
>
> Message: 1
> Date: Fri, 19 Jun 2009 10:
I don't have my settings to try it right now. Still i have a question. If
it's the way you describe it, why wouldn't sofia/extenal/f...@bar solve the
problem? I think i even copied the extension both to the default directory.
But i will confirm and let you know.
Max.
On Fri, Jun 19, 2009 at 4:55
On Fri, Jun 19, 2009 at 2:19 PM, Nik
Middleton wrote:
> You are indeed correct, it's the 64bit server that performs well, not
> the 32bit PAE version. I'm hoping that's the cause. I need to dig
> around and find out if it's possible to change the kernel remotely and
> see it sorts the issue. Ult
try removing certian elements of you setup to narrrow it down one at a time.
remove the lua + sql, the ivr scripts etc and see if you can pinpoint your
problem
it's amost for sure going to be in lua and probably some plugin for it.
On Fri, Jun 19, 2009 at 3:53 PM, Matthew Fong wrote:
> some l
Can you pastebin your script and your dialplan? There might be some clue
there..
On Fri, Jun 19, 2009 at 3:53 PM, Matthew Fong wrote:
> some lua event listeners are connecting to mysql with lua..but the
> connection is created once, and kept open
> the lua ivr's do *not *connect to any database.
some lua event listeners are connecting to mysql with lua..but the
connection is created once, and kept open
the lua ivr's do *not *connect to any database.
top -H seems to show an even distribution of of cpu and memory usage amongst
freeswitch threads. Nothing seems out of the ordinary with a spe
Now I feel stupid because I didn't read your original post closely enough.
You've defined your "mysocket" extension in the "public" context but when
you do an origination with sofia/internal/f...@bar it will use the "default"
context. I think the quickest way to handle this is to create a copy of
On another email thread Tony mentioned using "top -H" and then doing a gcore
to locate the process(es) that are sucking up all the resources. Have you
been down that path already?
-MC
On Fri, Jun 19, 2009 at 3:28 PM, Brian West wrote:
> Depends on what you're doing ... or not doing...
>
> /b
>
>
I'm not aware of any pre-existing software like this. FreeSWITCH has all the
hooks for someone to create the abstraction layers for CSTA, TAPI, VoiceXML,
etc. but no one has ponied up the money to pay for the development...
-MC
2009/6/18 Maxim Tsvetov
> Hello!
>
> We are seeking possibilities t
are you connecting to a db with the lua?
On Fri, Jun 19, 2009 at 3:20 PM, Matthew Fong wrote:
> With yesterday's trunk and also a release from 2 weeks ago, I noticed that
> my freeswitch process as it ran was eating up more and more memory. At the
> end of the day it was using 75% of the sever'
Depends on what you're doing ... or not doing...
/b
On Jun 19, 2009, at 3:20 PM, Matthew Fong wrote:
> With yesterday's trunk and also a release from 2 weeks ago, I
> noticed that my freeswitch process as it ran was eating up more and
> more memory. At the end of the day it was using 75% of
With yesterday's trunk and also a release from 2 weeks ago, I noticed that
my freeswitch process as it ran was eating up more and more memory. At the
end of the day it was using 75% of the sever's memory (About 12 gigs). It
starts out taking a small amount of memory, and then throughout the day it
That sounds like something it might do, but searching through the code I
didn't find any use of the flag that parameter sets. It is, of course,
possible that I missed it, especially if it's an indirect use.
2009/6/19 João Mesquita
> Inline...
>
> On Fri, Jun 19, 2009 at 3:27 PM, Bradley Brashie
Thanks!
On Fri, Jun 19, 2009 at 1:17 PM, Michael Collins wrote:
> Check out this change:
> http://fisheye.freeswitch.org/changelog/FreeSWITCH?cs=13505
>
> Possibly you need to wait for the CHANNEL_HANGUP_COMPLETE event?
> -MC
>
> On Fri, Jun 19, 2009 at 2:15 PM, Matthew Fong wrote:
>
>> I upgra
You are indeed correct, it's the 64bit server that performs well, not
the 32bit PAE version. I'm hoping that's the cause. I need to dig
around and find out if it's possible to change the kernel remotely and
see it sorts the issue. Ultimately I'll update it to 64 bit anyway, but
that's a 500 mile
Check out this change:
http://fisheye.freeswitch.org/changelog/FreeSWITCH?cs=13505
Possibly you need to wait for the CHANNEL_HANGUP_COMPLETE event?
-MC
On Fri, Jun 19, 2009 at 2:15 PM, Matthew Fong wrote:
> I upgraded to 13857 today, but noticed that the channel_hangup event no
> longer contain
I'm sorry, but like I said, I DONT WANT to do transcoding with FS(I
have another switch who is in charge of it, witch is from another
technology), I only want to negotiate the codecs in the way that I
want it. This only seams to work when bypass media or proxy media is
set to false. Due I n
FYI,
Just in case this affects you I wanted to give you all a heads up. With the
new NAT busting code we found some stuff in the default configs that was no
longer applicable and made some changes. Check them out:
http://fisheye.freeswitch.org/changelog/FreeSWITCH/?cs=13874
If you have any dialpl
Inline...
On Fri, Jun 19, 2009 at 3:27 PM, Bradley Brashier wrote:
> So it turns out that it wasn't a bug at all -- it is a feature that was not
> implemented. So I've got some work to do to get that running. Since I said I
> would, though, here's my analysis of the conference parameters you were
No right now you can not legally transcode G729 in FreeSWITCH, PERIOD!
/b
On Jun 19, 2009, at 2:11 PM, JuanMa wrote:
> Yes, it can do transcoding. Transcoding isn't the problem to my
> architecture, my problem is the codec negotiation between FS and
> Endpoints.
>
_
I upgraded to 13857 today, but noticed that the channel_hangup event no
longer contain the variable_billsec header.
Is this correct, or am I crazy? Thanks.
--matt
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freesw
Saeed Ahmad:
Yes, it can do transcoding. Transcoding isn't the problem to my
architecture, my problem is the codec negotiation between FS and
Endpoints.
I want to use FS as SBC (session border controller) or pseudo SIP
proxy. How i said in my last e-mail FS will work in bypass-media or
>mute-detect-sound
>Different sound for if muting using only when mute-detect flag is on.
Sorry... that didn't come out right. Try this:
Different sound to play when muted by someone else and mute-detect flag is
on. Plays mute-sound if this is not present.
On Fri, Jun 19, 2009 at 11:27 AM, Bradle
So it turns out that it wasn't a bug at all -- it is a feature that was not
implemented. So I've got some work to do to get that running. Since I said I
would, though, here's my analysis of the conference parameters you were
asking about:
mute-detect-sound
Different sound for if muting using only
Mike,
Here we go: http://pastebin.freeswitch.org/9447. Doesn't look very XML to me
though.
Max.
On Fri, Jun 19, 2009 at 2:10 PM, Michael Collins wrote:
> Max,
> that pastebin failed miserably as none of the xml shows up. can you try
> again or use our pastebin.freeswitch.org site?
> -MC
>
>
>
Max,
that pastebin failed miserably as none of the xml shows up. can you try
again or use our pastebin.freeswitch.org site?
-MC
On Fri, Jun 19, 2009 at 12:58 PM, Max Bridgewater wrote:
> Hi Mike,
>
> It's pasted here: http://pastebin.ca/1466521
>
> Thanks,
> Max.
>
>
>
>
> On Fri, Jun 19, 2009 a
Hi Mike,
It's pasted here: http://pastebin.ca/1466521
Thanks,
Max.
On Fri, Jun 19, 2009 at 11:43 AM, Michael Collins wrote:
> Can you turn on debugging (F8) and capture all the output after your
> originate? Put it into a pastebin. (pastebin.freeswitch.org)
> -MC
>
> On Fri, Jun 19, 2009 at 1
recovery_on_timer_expire was just my example..
I actually just want to try carrier B on everything except no_answer or
user_busy...
On Fri, Jun 19, 2009 at 6:06 AM, Brian West wrote:
> If you're getting RECOVERY_ON_TIMER_EXPIRE then you have maybe a NAT issue.
> /b
>
> On Jun 19, 2009, at 1:38 A
OK, I figured out the TTS stuff. It's a matter of choosing an engine (I
chose flite), uncommenting a few things, and setting the TTS variables in
the conference profile. You do have to rebuild FS to do this.
With that working, my count function works, too.
I posted a bug last night about conferen
Can you turn on debugging (F8) and capture all the output after your
originate? Put it into a pastebin. (pastebin.freeswitch.org)
-MC
On Fri, Jun 19, 2009 at 10:14 AM, Max Bridgewater wrote:
> Any help our there?
>
> I'm still trying to get this piece working. Essentially what i wan to do
> is,
On Fri, Jun 19, 2009 at 2:23 AM, Nik
Middleton wrote:
> I’m running Linux 2.6.18-128.1.10.el5PAE on i686 on the server, so sadly I
> don’t think that’s the issue
I could be wrong, but I think PAE is a 32-bit kernel adapted for
hardware with >4GB RAM. This can create a lot of overhead compared to
r
Any help our there?
I'm still trying to get this piece working. Essentially what i wan to do is,
when a call comes in (from registered devices as well as unregistered
devices), notify the my server socket. Somehow it's not working. The change
i made compared to the standard Freeswitch settings are
use top -H to get the per-thread cpu usage.
see if any one thread is using more than the rest.
then get a gcore of the running process and do a
thread apply all bt
and get a bt from the thread with the matching id.
Maybe that will tell you what is doing all the work.
On Fri, Jun 19, 2009 at 3:
Tim,
Look at lines 47 and 48 of the pastebin. I think something goofy is
happening there. What is 8...@x.x.x.x in your system? Is that the receive
fax extension?
-MC
-- Forwarded message --
From: Tim B
Date: Fri, Jun 19, 2009 at 7:39 AM
Subject: Re: [Freeswitch-users] Freeswitch-
If you're getting RECOVERY_ON_TIMER_EXPIRE then you have maybe a NAT
issue.
/b
On Jun 19, 2009, at 1:38 AM, Matthew Fong wrote:
the script is not part of a session or dial plan. :(
On Thu, Jun 18, 2009 at 11:31 PM, Jason White
wrote:
Mathieu Rene wrote:
> data="failure_causes=user_bus
here is the log...
http://pastebin.freeswitch.org/9440
haha, yeah i see it now... duh. pulled an all nighter, too many things going
on. must have overlooked it.
>
> When I connect to pastebin.freeswitch.org I get a helpful notice saying
> the login and password is pastebin/freeswitch
>
BTW, what is SWITCH? Can it do transcoding?
On Fri, Jun 19, 2009 at 2:06 PM, Saeed Ahmad wrote:
> It seems that you want to transcode G729 which is not possible. You can use
> it in passthru mode only.
>
>
> On Fri, Jun 19, 2009 at 1:48 AM, JuanMa wrote:
>
>> Hi,
>>
>> I need to have the habilit
It seems that you want to transcode G729 which is not possible. You can use
it in passthru mode only.
On Fri, Jun 19, 2009 at 1:48 AM, JuanMa wrote:
> Hi,
>
> I need to have the hability to negotiate the codec in a session (using
> proxy media or bypass media), unfortunally I've been unable to a
My freeswitch has a mysql database consists of freeswitch tables,
registrations and nibblebill on mysql configured it correctly and working...
Issue is when I call external ip's sometimes it works sometimes not?
2009-06-19 19:02:01 [INFO] switch_core_session.c:1040
switch_core_session_enable_hea
Tim B wrote:
>
> Michael, I ran the debugging you asked. I also tried to post it to
> pastebin.freeswitch.org but can't login. I used my login for the freeswitch
> site, but that doesn't seem to work?? How do I gain acess?
When I connect to pastebin.freeswitch.org I get a helpful notice sayin
Michael, I ran the debugging you asked. I also tried to post it to
pastebin.freeswitch.org but can't login. I used my login for the freeswitch
site, but that doesn't seem to work?? How do I gain acess?
Thanks.
Tim
> ---
I'm running Linux 2.6.18-128.1.10.el5PAE on i686 on the server, so sadly
I don't think that's the issue
Regards,
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Knight
Sent: 18
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