Hi Michael,
I did some tests but I haven't been successful, so there is what I'm
trying to achieve:
On A leg, my phone is using: PCMA and G729 (in this priority order)
With PEER A, I want to use only G729 (thats is the only codec that this
PEER support), so that the RTP flow will be:
Phone
See
http://wiki.freeswitch.org/wiki/Authoring_Freeswitch_Modules#Subscribing_to_events
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 2-Sep-09, at 8:47 AM, Mathieu Parent wrote:
> Hi,
>
> On Wed, Sep 2, 2009 at 4:37 PM, Br
Its because many phones cheat and just expect mwi without asking for it so
we send one on register.
There is an opt to disable it I think but I can't recall what it is atm
On Sep 2, 2009 9:20 PM, "mayamatakeshi" wrote:
On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi
wrote: > > Hello, > I'm test
On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi wrote:
> Hello,
> I'm testing FS support for the header Path (FS is behind opensips).
> It pretty much works: I tested calling from one user to the other and calls
> work perfectly.
> However, I've noticed that when I register my terminal directly with
Hey guys,
Thanks for all the great replies. After seeing all the people who've
gotten it working and configured, I feel pretty confident that if we
go this path, we'll be successful.
I can't say too much about the app, but in essence we just need to
take in traffic over a E1 connection a
I will put several nickels saying it is impossible :)
seriously, can it be done?
T.
On Wed, Sep 2, 2009 at 10:35 PM, Tim Meade wrote:
> I am very interested in a response to this. Last I knew there was only
> T.38 pass through and for some reason I'm not even sure it that was fully
> implem
I am very interested in a response to this. Last I knew there was only T.38
pass through and for some reason I'm not even sure it that was fully
implemented.
Tim
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Tihomir
Cu
Hello Eric,
I just interconnected FS/openzap and Panasonic via E1 PRI trunk (euroisdn)
using libpri stack, and it works just fine.
Feel free to drop on irc for any help.
Regards,
Ognjen
On Wed, Sep 2, 2009 at 7:29 PM, Michael Collins wrote:
>
>
> On Wed, Sep 2, 2009 at 8:52 AM, Eric Richmond
well your clock shouldn't be going back in time... that is unless you
have figured out time travel or passed thru some star trekish temporal
wake.
For the most part its a harmless warning unless its happening every
second or so.
/b
On Sep 2, 2009, at 2:43 PM, Phillip Jones wrote:
> Hi th
Her real name is Katherine
/b
On Sep 2, 2009, at 2:27 PM, Carlos S. Antunes wrote:
> http://www.gmvoices.com
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSC
Hi there,
Can anyone give any insight to this following message:
2009-09-02 15:21:44.665608 [CRIT] switch_time.c:454 Reverse Clock Skew
Detected!
This is on a WIN2003 machine with the last call hangup exactly 20 minutes
and 20 seconds earlier.
Just wondering how CRITICAL this really is?
Many t
Hi guys,
just a quick question... is it possible to do a reliable on the fly T30 <>
T38 transcoding at all ... what is the status of T.38 on FS ?
T,
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mail
On Wed, Sep 2, 2009 at 12:14 PM, Nick Lemberger wrote:
> I'm guessing she's a Cepstral voice, but can I ask what version, khz and
> settings the 8khz sounds are recorded with (I ask about the khz because
> perhaps they were pre-transcoded)? I tried downloading 5.1 from Cepstral
> but they don't s
I have a production application where I use FS as part of small, custom ACD
solution, with about 80 incoming DIDs and 4 agent positions. It's been
deployed for about 4 months now, and was in beta long before that... So
far, excellent perfomrance on Windows 2003 server, 32-bit, with 4GB of
memory.
http://www.gmvoices.com
She appears to be a real person! :)
Nick Lemberger wrote:
> I'm guessing she's a Cepstral voice, but can I ask what version, khz and
> settings the 8khz sounds are recorded with (I ask about the khz because
> perhaps they were pre-transcoded)? I tried downloading 5.1 fr
Taken from a similar recent posting:
"Callie" is one of the voices from GM Voices. She is definitely available
for custom work. Visit www.gmvoices.com for more info. Tell them that the
FreeSWITCH project sent you. :) MC
On Wed, Sep 2, 2009 at 3:14 PM, Nick Lemberger wrote:
> I'm guessing she's a
I'm guessing she's a Cepstral voice, but can I ask what version, khz and
settings the 8khz sounds are recorded with (I ask about the khz because perhaps
they were pre-transcoded)? I tried downloading 5.1 from Cepstral but they
don't sound at all alike. I'd like to replace a few prompts with wo
Good to know, I am getting Sangoma cards for my FreeSWITCH as well.
Diego
On Wed, Sep 2, 2009 at 5:29 PM, Michael Collins wrote:
>
>
> On Wed, Sep 2, 2009 at 8:52 AM, Eric Richmond wrote:
>
>> (Sorry if this has been posted before, or even 2x. I sent it out the
>> first time with the wrong em
yikes, dont swig your own stuff
just delete all the files that had merge issues and update again.
On Wed, Sep 2, 2009 at 1:14 PM, Peter P GMX wrote:
> I had the same problem. Must have been changed something in lua since
> this morning.
>
> Please install swig.
>
> E.g. on Debian
> sudo apt-ge
I had the same problem. Must have been changed something in lua since
this morning.
Please install swig.
E.g. on Debian
sudo apt-get install swig
That did it for me.
Best regards
Peter
Lars Zeb schrieb:
>
> I just updated using “svn up” which brought the source to 14741. After
> running “./con
I just updated using "svn up" which brought the source to 14741. After
running "./configure", I ran "make" and got the following output:
making all mod_lua
make[5]: swig: Command not found
make[5]: *** [mod_lua_wrap.cpp] Error 127
make[4]: *** [all] Error 1
make[3]: *** [mod_lua-all] Error
On Wed, Sep 2, 2009 at 8:52 AM, Eric Richmond wrote:
> (Sorry if this has been posted before, or even 2x. I sent it out the
> first time with the wrong email address, and the 2nd time before i
> confirmed I wanted to be on the list, so my assumption is that it
> hasn't gone out yet, although I m
Thanks for the replies guys, its much appreciated.
On Sep 2, 2009, at 12:52 PM, Moises Silva wrote:
On Wed, Sep 2, 2009 at 11:52 AM, Eric Richmond wrote:
Hello all,
I'm wondering if anyone has successfully deployed a freeswitch server
with an E1 card attached. By looking at the wiki and chat
On Wed, Sep 2, 2009 at 11:52 AM, Eric Richmond wrote:
> Hello all,
>
> I'm wondering if anyone has successfully deployed a freeswitch server
> with an E1 card attached. By looking at the wiki and chatting in IRC,
> I've heard that theoretically a FS + Sangoma E1 card solution should
> work, but
Eric,
We have tried it, and it worked after a few tweaks on our PRI line,
using the standard openzap plus wanpipe driver of Sangoma, the wiki on
Sangoma site is old, but the one on freeswitch kinda works.
Thanks & Regards,
Mitul Limbani,
Founder & CEO,
Enterux Solutions Pvt. Ltd.,
The Enterpr
(Sorry if this has been posted before, or even 2x. I sent it out the
first time with the wrong email address, and the 2nd time before i
confirmed I wanted to be on the list, so my assumption is that it
hasn't gone out yet, although I might be wrong, and if it has gone out
before, I'm sorry
Hello all,
I'm wondering if anyone has successfully deployed a freeswitch server
with an E1 card attached. By looking at the wiki and chatting in IRC,
I've heard that theoretically a FS + Sangoma E1 card solution should
work, but I'd really like to hear from someone who actually has it
wo
Hi,
On Wed, Sep 2, 2009 at 4:37 PM, Brian West wrote:
> Using esl + perl you could do it.
>
> /b
>
On Wed, Sep 2, 2009 at 4:46 PM, Diego Viola wrote:
> Or you can do something like this with Ruby + FSR:
>
> http://pastebin.freeswitch.org/10184
>
thanks for your suggestions, but this also require
i mean valgrind is very intensive so you must run very slow 1-5cps
yes if you have a version that only has log-file you can use that.
if you find me on irc and send me the credentials privately I will examine
your box for you.
On Wed, Sep 2, 2009 at 10:26 AM, Benedikt Fraunhofer <
fraunhofer.li
2009/9/2 Michael Collins :
> Are you trying to get a channel variable or capture DTMF input from the
> caller?
i try to make IVR by php outbound socket. in XML dialplan we can get
DTMF by read application (store in channel variable)
I found it's success in perl outbound (IVR.pm) but for php how do
That's what I ended up with earlier today. Transfering to another
extension that does the att_xfer/bridge and rebinds meta app. I think
it works. Unfortunately my third phone is out of power, so haven't had
much chance to test it.
2009/9/2 Anthony Minessale :
> instead of bridge or att_xfer then u
w00t!
On Wed, Sep 2, 2009 at 6:44 AM, Helmut Kuper wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hello,
>
> just to close this thread, I would like to say that after 21000
> pstn-calls and 45 days up time (old stack: max 10 days) the ISDN call
> resource management based on Q931 t
Great stuff, thanks for your hard work :).
Keep it up.
Regards,
Diego
On Wed, Sep 2, 2009 at 1:44 PM, Helmut Kuper wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hello,
>
> just to close this thread, I would like to say that after 21000
> pstn-calls and 45 days up time (old stac
Are you trying to get a channel variable or capture DTMF input from the
caller?
-MC
On Wed, Sep 2, 2009 at 7:56 AM, Dome Charoenyost wrote:
> I follow
> http://wiki.freeswitch.org/wiki/PHP_ESL#ivrd
>
> how to get from php ?
>
>
> Dome C.
> --
>
Hello Anthony,
2009/9/2 Anthony Minessale :
> run it slower and make sure it shuts down clean.
i already reduced load 37% but that didnt help, now i'm down to 25%
and it's running.
> valgrind --tool=memcheck --log-file-exactly=vg.log --leak-check=full
> --leak-resolution=high --show-reachable=ye
instead of bridge or att_xfer then use transfer to transfer to an extension
that does the bridge.
or transfer to the inline dialplan.
On Wed, Sep 2, 2009 at 5:00 AM, Harry Vangberg wrote:
> Ah. att_xfer seems nice. But, it still doesn't allow C to eventually
> rebridge A to B (or possibly D, E
run it slower and make sure it shuts down clean.
valgrind --tool=memcheck --log-file-exactly=vg.log --leak-check=full
--leak-resolution=high --show-reachable=yes /path/to/freeswitch -vg
On Wed, Sep 2, 2009 at 10:11 AM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:
> You have to reduce
I'm planning to deploy on windows a small call center (around 50 people) and
willing to help anyhow. I will be able to test on machines mounting windows
2003 server. Is there a standard test that could be employed to correctly
benchmark the results?
On Wed, Sep 2, 2009 at 15:50, Brian West wrote:
You have to reduce the load when running valgrind.
On Wed, Sep 2, 2009 at 7:59 AM, Benedikt Fraunhofer <
fraunhofer.lists.freeswitch-...@traced.net> wrote:
> Hello *,
>
> 2009/8/31 Rupa Schomaker :
> > Isn't there a known issue with lua+sql leaking memory on some platforms?
>
> just lua, no sql
Woof!
On Tue, 01 Sep 2009 18:52:01 -0400, Anthony Minessale
wrote:
> there is no chance that you would not enter the conf muted the way you
> describe unless you are using an older revision of FS that had a bug in
> the parsing of the conference flags.
Perhaps some listeners are hitting the
sip_hangup_disposition variable
On Wed, Sep 2, 2009 at 4:42 AM, lakshmanan wrote:
>
>
> I want to find which leg has cut the call? Is it possible to find that in
> freeswitch?
> For example:
> A and B are speaking. If B cuts the call, then I need to play appropriate
> message to A, and if A cuts
I follow
http://wiki.freeswitch.org/wiki/PHP_ESL#ivrd
how to get from php ?
Dome C.
--
#!/usr/bin/php -q
2009/9/2 Brian West :
> uuid_getvar
>
> /b
>
> On Sep 2, 2009, at 8:16 AM, Tristan Mahé wrote:
>
>> Hi,
>>
>> just a fast 2cent:
>>
>>
Or you can do something like this with Ruby + FSR:
http://pastebin.freeswitch.org/10184
On Wed, Sep 2, 2009 at 2:27 PM, Mathieu Parent wrote:
> Hi,
>
> I wanted to run an "originate" command when a MESSAGE_WAITING event is
> fired.
>
> Is there a simpler way than creating a daemon listening to t
As far as we'll solve all FS configuration issues I think we'll make some
tests on Windows platform and will share results with community.
Please if someone has an opportunity to test FS on modern windows server
editions (32/64 bit) - submit your test results too. It is interesting to
put together
Using esl + perl you could do it.
/b
On Sep 2, 2009, at 9:27 AM, Mathieu Parent wrote:
> Hi,
>
> I wanted to run an "originate" command when a MESSAGE_WAITING event
> is fired.
>
> Is there a simpler way than creating a daemon listening to the event
> socket ?
>
> Thanks
>
> Mathieu Parent
Hi,
I wanted to run an "originate" command when a MESSAGE_WAITING event is fired.
Is there a simpler way than creating a daemon listening to the event socket ?
Thanks
Mathieu Parent
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.
Hello Brian,
2009/9/2 Brian West :
> What are you doing in these lua scripts? Because there are a few things you
> can do in the lua script itself that will cause you to leak like crazy due
> to improper use.
> /b
the setup is the same as in http://jira.freeswitch.org/browse/MODSOFIA-22
one is
I know people that have deployed on windows... not a huge problem just
hasn't been load tested like linux... we don't have the resources or
time to load test every single platform, tune and tweak it. The
community can help out with this area a lot.
/b
On Sep 2, 2009, at 8:01 AM, Diego Tor
uuid_getvar
/b
On Sep 2, 2009, at 8:16 AM, Tristan Mahé wrote:
> Hi,
>
> just a fast 2cent:
>
> get var via channel status ? ( variable_res )
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
just to close this thread, I would like to say that after 21000
pstn-calls and 45 days up time (old stack: max 10 days) the ISDN call
resource management based on Q931 timers delivered by the *new* Q931
stack is still clean. Each done call was
What are you doing in these lua scripts? Because there are a few
things you can do in the lua script itself that will cause you to leak
like crazy due to improper use.
/b
On Sep 2, 2009, at 7:59 AM, Benedikt Fraunhofer wrote:
Hello *,
2009/8/31 Rupa Schomaker :
Isn't there a known issue
You don't have to do that anymore... the default profile on 5060 will
work when the users are on the public internet also. No need to have
two profiles anymore.
/b
On Sep 2, 2009, at 7:50 AM, Juan Backson wrote:
> Hi,
>
> Things are working find before I tried using public IP ( behind
> N
Hi,
just a fast 2cent:
get var via channel status ? ( variable_res )
Dome Charoenyost a écrit :
> Dear sir,
>
> How to get digit from outbound php esl ?
> example in my php
>
>echo "sendmsg\n";
>echo "call-command: execute\n";
>echo "execute-app-name: read\n";
>echo "exec
What is the reason for saying this? Perhaps the effort of the development group
of FS has been wasted trying to support Windows as a platform for production
systems?
Diego
http://lacarretade.blogspot.com/
--- On Tue, 9/1/09, Muhammad Shahzad wrote:
From: Muhammad Shahzad
Subject: Re: [Frees
Hello *,
2009/8/31 Rupa Schomaker :
> Isn't there a known issue with lua+sql leaking memory on some platforms?
just lua, no sql in use :)
> On Mon, Aug 31, 2009 at 8:32 AM, Brian West wrote:
>> Use valgrind.
i tried that... in the beginning valgrind segfaultet several times
with some error mess
Hi,
Things are working find before I tried using public IP ( behind NAT ) to
register IP phones. I am getting:
2009-09-02 20:46:50.575837 [WARNING] sofia_reg.c:1713 Can't find user
[180...@public-ip]
You must define a domain called 'public-ip' in your directory and add a user
with the id="18000
Dear sir,
How to get digit from outbound php esl ?
example in my php
echo "sendmsg\n";
echo "call-command: execute\n";
echo "execute-app-name: read\n";
echo "execute-app-arg: 0 20
/opt/freeswitch/sounds/th/tuxza/welcome.wav res 5000 #\n\n";
How to get res ?
best regards.
D
Hello,
I'm testing FS support for the header Path (FS is behind opensips).
It pretty much works: I tested calling from one user to the other and calls
work perfectly.
However, I've noticed that when I register my terminal directly with FS
without going thru the proxy, I receive an unsolicited NOTIF
Ah. att_xfer seems nice. But, it still doesn't allow C to eventually
rebridge A to B (or possibly D, E etc) at some point in the
conversation, where the caller needs to talk with somebody else.
2009/9/1 Anthony Minessale :
> you probably don't want to call bridge from bind meta app, try using the
I want to find which leg has cut the call? Is it possible to find that in
freeswitch?
For example:
A and B are speaking. If B cuts the call, then I need to play appropriate
message to A, and if A cuts the call, I need to play some messages to B.
What is the way to do this?
--
View this message
60 matches
Mail list logo
