FYI, I opened http://jira.freeswitch.org/browse/MODLANG-97 on this issue.
-MC
On Fri, Jan 16, 2009 at 9:03 AM, Michael Jerris wrote:
> All long running non js code should be wrapped in the suspend/resume gc
> stuff. For example:
> cb_state.ret = BOOLEAN_TO_JSVAL(JS_FALSE);
> cb_state.saveDepth =
Andrew,
On behalf of the OSS telephony community, and particularly the
FreeSWITCH community, many thanks to you! We appreciate all the stuff
you've given back. Thanks for showing the true Open Source spirit!
-MC
On Fri, Jan 16, 2009 at 3:35 PM, Andrew Thompson wrote:
> I'd like to announce the f
I'd like to announce the first beta release of a cross-platform ruby/tk
GUI for using FreeSWITCH like a soft-phone (using mod_portaudio). It's
not particularly fancy, but I needed a cross platform softphone with
good voice quality that was debuggable and didn't have a ton of
features to confuse the
I would suspect the NAT wasn't punching holes or lied. :)
/b
On Jan 16, 2009, at 3:57 PM, Will Smith wrote:
Well, if NAT involved, why did I get through after I put the call on
hold and take the call back. I am getting the SIP trace, hope that
will show something.
Thank you all
Well, if NAT involved, why did I get through after I put the call on hold and
take the call back. I am getting the SIP trace, hope that will show something.
Thank you all
--- On Fri, 1/16/09, Brian West wrote:
From: Brian West
Subject: Re: [Freeswitch-users] Dialing Out Problem via Gateway
To:
NAT involved?
/b
On Jan 16, 2009, at 3:30 PM, Will Smith wrote:
Thank you Brian,
The problem is very simple, I or the other party cannot hear each
other when I first dial and the other party picks up the phone. We
hear the phone ring, the other end picks up the phone says
something, but
This is a known issue with all virtualization solutions. The realtime
clocks inside the Virtual Machine jitter quite a bit which causes
havoc with the udp media streams. I have never heard of someone using
any VoIP product inside a VM and being happy with the result. To the
best of my knowledge, al
Thank you Brian,
The problem is very simple, I or the other party cannot hear each other when I
first dial and the other party picks up the phone. We hear the phone ring, the
other end picks up the phone says something, but I cannot hear - nothing, even
static. Same thing happen on my end, I
A SIP trace would be extremely helpful.
http://wiki.freeswitch.org/wiki/Troubleshooting_Freeswitch#Enabling_SIP.2FSofia_Tracing
-MC
On Fri, Jan 16, 2009 at 1:13 PM, Brian West wrote:
> Can you detail your problem a bit more?
> /b
> On Jan 16, 2009, at 3:09 PM, Will Smith wrote:
>
> Hi,
> I got a
Can you detail your problem a bit more?
/b
On Jan 16, 2009, at 3:09 PM, Will Smith wrote:
Hi,
I got a strange problem that I don't really understand, and I hope
that you could give me some hint how to fix that:
When I dial out through a gateway that is defined in the
sip_profiles/externa
Hi,
I got a strange problem that I don't really understand, and I hope that you
could give me some hint how to fix that:
When I dial out through a gateway that is defined in the
sip_profiles/external , (The xml file is simple as below. ) I cannot talk or
hear from the other end. But when I pu
I have chown the freeswitch directory to my user imyrvold, therefore
I put it in ~/Library/LaunchDaemons.
Do you run freeswitch as root, as you put it in /System/library/
LaunchDaemons? That directory should be reserved anyway for Apple's
system tools.
A better idea would be to put it in /Lib
On Jan 16, 2009, at 5:45 AM, Shido Xavier wrote:
> Please specify Intel or PPC.
Very good point and I had had that thought as well.
I am on PPC 10.4.11.
Thanks for any help or ideas.
Marty
>
>
> -Greg M.
>
>
> On Fri, Jan 16, 2009 at 1:29 AM, Martin Joseph
> wrote:
>>
>> On Jan 15, 2009, a
On Jan 16, 2009, at 8:09 AM, Ivan C Myrvold wrote:
> I haven't tried using launchctl for FreeSWITCH. But when I saw your
> post, I tried it out. I have no problem getting it to work:
>
> I make a file "org.freeswitch.freeswitch.plist" and save it to ~/
> Library/LaunchAgents with the following co
Yes it's hard to trust virtualized stuff because you have no idea what they
skimp on in terms of realtime access.
I won't endorse using FS on a VM as i have not done it very extensively
beyond openvz but I can
point out a few reasons why it has a fighting chance.
FS uses a timer architecture desi
Hi
This is what I found in the manual
SMS and MMS
The GSM and CDMA modules can send and receive SMS text messages and
MMS multimedia messages to or from a mobile phone. These are
transmitted over the Ethernet port as SIP MESSAGE messages to ensure
compatibility with a variety of different SIP PBX
all gateways are properties of a profile
so even in the parse domain way of handling gateways whatever profile parsed
the domain
will be the owner of all the gateways discovered from the domain.
FYI,
The internal profile has nothing to do with users or anything else.
You are confusing the default
All long running non js code should be wrapped in the suspend/resume
gc stuff. For example:
cb_state.ret = BOOLEAN_TO_JSVAL(JS_FALSE);
cb_state.saveDepth = JS_SuspendRequest(cx);
args.input_callback = dtmf_func;
args.buf = bp;
args.buflen = len;
Please specify Intel or PPC.
-Greg M.
On Fri, Jan 16, 2009 at 1:29 AM, Martin Joseph wrote:
>
> On Jan 15, 2009, at 3:10 PM, Michael Jerris wrote:
>
>> Your build issue is with your autotools install, I have seen issues if
>> you have ever installed any of the autotools from macports or fink.
>
I haven't tried using launchctl for FreeSWITCH. But when I saw your
post, I tried it out. I have no problem getting it to work:
I make a file "org.freeswitch.freeswitch.plist" and save it to ~/
Library/LaunchAgents with the following content:
http://www.apple.com/DTDs/PropertyList-1.0.dtd
">
Tamas,
The channel variable won't work for you if you can't ignore early
media. Your best bet is to use the variable execute_on_answer to
transfer an answered call to a new extension. Then you could just
sleep for 15sec and then check the value of endpoint_disposition.
What is the applicati
Raymond Chandler wrote:
> Apostolos Pantsiopoulos wrote:
>> When I am using the following method to place a call from the dialplan :
>>
>> sofia/gateway//
>>
>> how do I tell FS which profile to use (as in the
>> sofia// method?)
>>
>> I am asking that because all my calls to my declared use
>>
Please do not report bugs on the mailing list.
It's very hard to keep track of them this way.
Please file all bugs to jira so we will not lose track of them.
http://jira.freeswitch.org
On Fri, Jan 16, 2009 at 1:25 AM, Helmut Kuper wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hel
Apostolos Pantsiopoulos wrote:
When I am using the following method to place a call from the dialplan :
sofia/gateway//
how do I tell FS which profile to use (as in the
sofia// method?)
I am asking that because all my calls to my declared use the
5080 port,
and I want them to use the 5060
Hello,
It seems originate_timeout isn't take effect when we got early media.
Our carrier is sending the ring tone in early media, so if I try
timeout isn't occur after 15s
however with:
timeout is OK. but we don't get early media, I think it would be nice to
not ignore it
I found ring_ready
When I am using the following method to place a call from the dialplan :
sofia/gateway//
how do I tell FS which profile to use (as in the
sofia// method?)
I am asking that because all my calls to my declared use the
5080 port,
and I want them to use the 5060 port. Is there a way to configure
I've found the problem. one js thread wait in socket.read
(mod_spidermonkey_socket) on data.
That caller have hangup, which means that the garbage collector waits on it
to close.
All new javascript sessions waits in JS_AWAIT_GC_DONE for the garbage
collector to be done before proceeding (which mea
Any news regarding this issue?
Apostolos Pantsiopoulos wrote:
I am attaching the wireshark capture. Openphone is on xxx.xxx.xxx.202 and
FS is on xxx.xxx.xxx.212
Robert Jongbloed wrote:
Can you send me a WireShark capture?
Robert Jongbloed
OPAL/OpenH323/PTLib Architect and Co-founder.
I've got a loop, but the first thing checked in each iteration is if
session.ready() returns false (and in that case exit the loop).
I do create sessions in the script: create, try to originate to a
destination and then finally bridge together the caller and the new session.
I'll try to give you m
Speaking of networking...
After timing that's the next "achilles heel" of RTP handing with virtualization.
Very, very few of these platforms were designed to handle massive
numbers of very small RTP packets. Everything from interrupt handling
on the actual ethernet adapter to getting the data in
30 matches
Mail list logo
