Anthony, thank you very much for your response. The daemon that was
reading the events froze, so apparently that was the source of the
problem and your explanation fits perfectly.
On Mon, Dec 28, 2009 at 12:47 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
most likely cause would be
I just got into the fs cli and when I ran a 'show calls' I got the
following message:
2009-12-24 09:58:20.058365 [CRIT] mod_event_socket.c:337 Lost 8456 events!
What does this mean? does it mean the event_socket did not report 8456
events? Why could this happen?
The answer to this is pretty
Hi, I wanted to know what is the javascript equivalent of lua's
os.execute(). I need to run a command from within a js script.
Thanks!
Nicolas
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
I'm looking for the equivalent configuration parameter or option of
Asterisk's canreinvite
(http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite). Is
there anything like this for configuring a gateway? (there's no info
about it on the wiki).
Thanks!
Nicolas
Thanks, but I would like to keep FS in the media path. What would be
the equivalent of an Asterisk sip.conf's canreinvite=no?
On Tue, Dec 15, 2009 at 1:45 PM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
Closest thing I've found:
Thanks!
On Tue, Dec 15, 2009 at 3:12 PM, Frank Carmickle fr...@carmickle.com wrote:
On Tue, Dec 15, Nicolas Brenner wrote:
Thanks, but I would like to keep FS in the media path. What would be
the equivalent of an Asterisk sip.conf's canreinvite=no?
It's that way by default. Fs wants
I had a voip provider which wouldn't accept calls from Freeswitch because of
the user-agent string. I had to change it to Asterisk and then everything
worked.
Nico
On Wed, Nov 18, 2009 at 12:21 PM, Brian West br...@freeswitch.org wrote:
you do realize that is NOT the purpose of the user-agent
it doesn't assume auth it just calculates the response
hash differently on this case where qop isn't present.
/b
On Oct 6, 2009, at 4:22 PM, Nicolas Brenner wrote:
What does the qop parameter stand for? Apparently because of that
parameter, FS sends a new REGISTER including
... what purpose would auth serve in the
first place?
/b
On Oct 7, 2009, at 8:48 AM, Nicolas Brenner wrote:
Is there some way to make FS register with the gateway that is
rejecting the authentication? is it FS or the SIP server at fault?
Why would X-Lite work and FS not?
Thanks again
messing with packets... notice it says
rport 5080 but we are sending to 5060.
/b
On Oct 5, 2009, at 11:42 PM, Nicolas Brenner wrote:
Ignore my previous email, the traces were incomplete, got much better (and
complete) traces with ngrep (found a suggestion from Brian in the list
archive
a pcap dump as with the
siptrace feature on the cli?
On Mon, Oct 5, 2009 at 12:20 PM, Michael Collins m...@freeswitch.org wrote:
On Sun, Oct 4, 2009 at 4:09 PM, Nicolas Brenner nico...@medularis.comwrote:
Mike, how exactly should I format the file? I got the pcap file, how do I
convert
Thanks again for your time and help!
Nicolas
On Tue, Oct 6, 2009 at 12:19 AM, Nicolas Brenner nico...@medularis.comwrote:
There was no sane way of doing that, so I ended up logging the trace from
the cli.
Here's the bad registration:
- http://pastebin.freeswitch.org/10605
Here's
:29 PM, Nicolas Brenner wrote:
Any ideas about this?
The SIP provider is offering H323, but I'm not quite sure about that, is
mod_opal working right?
Thanks!
Nicolas
On Tue, Sep 29, 2009 at 6:42 PM, Nicolas Brenner nico...@medularis.comwrote:
Anthony, thanks. Below are my config files
? Ends up being too much noise so I just don't bother.
Mike
On Oct 4, 2009, at 6:19 PM, Nicolas Brenner wrote:
Here it is:
- http://pastebin.freeswitch.org/10582
(it is the pcap file I sent on the first email of this thread, converted to
text with 'tshark -V -r')
On Sun, Oct 4, 2009 at 5:40
Any ideas about this?
The SIP provider is offering H323, but I'm not quite sure about that, is
mod_opal working right?
Thanks!
Nicolas
On Tue, Sep 29, 2009 at 6:42 PM, Nicolas Brenner nico...@medularis.comwrote:
Anthony, thanks. Below are my config files for the two gateways from the
sip
Just finished downloading the whole torrent and seeding now.
On Tue, Sep 29, 2009 at 3:04 PM, Dan White dwh...@olp.net wrote:
I've downloaded 0Kb in 56 minutes. Anyone mind setting up a seeder? I'm on
a pretty fast connection.
While waiting, I found the following discussions:
I found the
Hello everyone,
I am trying to add a gateway, but after configuring it just like the others
gateways I have, it is failing to register with a message like this:
2009-09-29 12:54:40.853440 [ERR] sofia_reg.c:1402 orange Registration Failed
with status Operation has no matching challenge [904].
with XXX maybe we
can see the issue for you.
On Tue, Sep 29, 2009 at 1:05 PM, Nicolas Brenner nico...@medularis.comwrote:
Hello everyone,
I am trying to add a gateway, but after configuring it just like the
others gateways I have, it is failing to register with a message like this:
2009-09-29
Did a little more digging, ALLOTTED_TIMEOUT has an error code of 602
according to the Wiki (http://wiki.freeswitch.org/wiki/Hangup_causes)
nevertheless that code is not covered in RFC 4497 (
http://tools.ietf.org/html/rfc4497)
On Mon, Sep 21, 2009 at 8:41 PM, Nicolas Brenner nico
Hi,
Today, while trying to bridge some calls I started to get a ALLOTTED_TIMEOUT
hangup cause on the second leg. I looked for info on the Wiki and Google,
but I couldn't find a detailed explanation. Does anybody know what does it
mean exactly?
Thanks!
Nicolas
I gave up on compiling esl, I got a bunch of errors, there were several
people on the list with problems and apparently no straight solution,
especially for php-esl. I am now using a ruby library, posted here by Diego
Viola I believe.
On Tue, Sep 1, 2009 at 2:33 AM, Michael Collins
Hi, I'm trying to get dtmf input, but I'm not getting anything.
What I discovered though, is that my provider is at fault, since when I
switched to another voip provider, everything started to work beautifully.
My question is: since my provider is not doing RC2833 dtmf (even though they
say they
a écrit :
On Fri, Aug 07, 2009 at 06:10:25PM -0400, Nicolas Brenner wrote:
Hi,
I'm trying to get started with the ESL using PHP. I compiled the ESL, then
phpmod according to the wiki instructions, but then when I try the examples
in the libs/esl/php dir, they fail saying:
PHP Fatal error
at 3:45 PM, Nicolas Brenner nico...@medularis.comwrote:
Hi Matt,
Actually I'm explicitly setting hangup_after_bridge to true, think setting
it to false would help? I'm going to try that.
Here's the JS code:
(Note: session.getVariable() doesn't work, FS complains saying it is not a
function
pjinthe...@gmail.com wrote:
What does
bridge_hangup_cause
give you?
On Fri, Aug 7, 2009 at 12:43 PM, Nicolas Brenner nico...@medularis.com
wrote:
I changed the script to set hangup_after_bridge to false, but still the
same thing happens, I get this on the console:
2009-08-07 12:27
Hi,
I'm trying to get started with the ESL using PHP. I compiled the ESL, then
phpmod according to the wiki instructions, but then when I try the examples
in the libs/esl/php dir, they fail saying:
PHP Fatal error: Cannot redeclare ESLconnection::__construct() in
I'm bridging 2 calls in a javascript file, I originate the first call and
then execute a bridge with an origination string for the second call. If I
hangup the first call while trying to make the second call, I get this on
the console:
2009-08-05 16:44:05.69122 [NOTICE]
://www.hellohunter.com
On Thu, Aug 6, 2009 at 9:38 AM, Nicolas Brenner nico...@medularis.comwrote:
I'm bridging 2 calls in a javascript file, I originate the first call and
then execute a bridge with an origination string for the second call. If I
hangup the first call while trying to make
I'm bridging 2 calls in a javascript file, I originate the first call and
then execute a bridge with an origination string for the second call. If I
hangup the first call while trying to make the second call, I get this on
the console:
2009-08-05 16:44:05.69122 [NOTICE]
the trick
?
Not sure if/how “sched_broadcast” functions when the call has not yet been
bridged though…
Let us know..
Best Regards
Keith
*From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Nicolas
Brenner
*Sent
Hi,
I have a small JS script that calls a phonenumber, when the call is answered
it plays a wave file, then it calls a second phonenumber and bridges the
calls. Is it possible to make wave-playing async, so that the second call is
generated as soon as the first is picked up? Right now the wave
Matthew, Anthony and Michael, thank you very much, seems like you gave me
exactly the info I needed!
On Thu, Jul 30, 2009 at 4:25 PM, Michael Collins m...@freeswitch.org wrote:
On Thu, Jul 30, 2009 at 12:36 PM, Nicolas Brenner
nico...@medularis.comwrote:
Thanks, I'll try that.
How can
I would like to originate 2 calls from FS and then bridge them. There's no
incoming call so I think there's no dialplan involved.
What I'd like to do now is apply lcr rules to these calls. I've come up with
2 options so far:
1) call lcr through the socket twice (once for each phonenumber) and
with:
originate ${lcr_auto_route} extension
extension just needs to match something in your dialplan.
In extension, you'd do another lcr lookup and then bridge to that leg's
${lcr_auto_route} value.
On Tue, Jul 21, 2009 at 10:35 AM, Nicolas Brenner
nico...@medularis.comwrote:
I would
:
originate lcr_auto_route1 bridge(lcr_auto_route2)
How soon do you need this?
On Tue, Jul 21, 2009 at 11:27 AM, Nicolas Brenner
nico...@medularis.comwrote:
That looks like a good way to go about it.
How can I access channel variables through the socket using the api? I
mean, how do I recover
route. Then when that call establishes, it'll hit the dialplan with the
second number which will also be routed through lcr.
Is that more what you are looking for?
This way all the 'routing' logic can be done via the dialplan.
On Tue, Jul 21, 2009 at 1:00 PM, Nicolas Brenner nico
Great! Thanks!
On Tue, Jul 21, 2009 at 2:51 PM, Rupa Schomaker r...@rupa.com wrote:
Just a note that the as xml syntax has been added to current trunk.
On Tue, Jul 21, 2009 at 1:21 PM, Rupa Schomaker r...@rupa.com wrote:
Well, the as xml is something I've been meaning to do, so I'm gonna
Hi,
Today I ran out of credit in one of my voip providers. When this happened,
all my outgoing calls started failing with hangup cause
NETWORK_OUT_OF_ORDER. Once I got some more credit, the calls kept failing. I
restarted freeswitch and then everything worked fine again.
Unfortunately this is
something to do with NAT, and in this case there's no
NAT involved. Anyway, I'm running a rev 13973 so I'll update to the latest
svn rev and hope it doesn't happen again.
On Fri, Jul 17, 2009 at 4:35 PM, Nicolas Brenner nico...@medularis.comwrote:
Hi,
Today I ran out of credit in one of my voip
I have a small JS script that makes a call, plays a sound file and then
hangs up. For each call it makes, I log the hangup_cause variable on the
CHANNEL_HANGUP_COMPLETE event. Most of the time, when calls are successful,
I get a NORMAL_CLEARING cause, but sometimes I'll get a NONE cause. I wanted
Thank you for all the patience and effort. You've done a great work! Have a
great meal!
On Thu, Jun 18, 2009 at 12:48 PM, Michael Collins m...@freeswitch.orgwrote:
Thank you so much! The devs are really loving this.
-MC
On Thu, Jun 18, 2009 at 11:40 AM, Saeed Ahmad
...@gmail.com wrote:
On Mon, Apr 20, 2009 at 3:10 PM, Nicolas Brenner nico...@medularis.com
wrote:
Hi,
I might be in a position (finally) to ask/suggest one of my voip
providers
to use an alternative codec to G729. I wanted to know what would be the
best
replacement for it.
Thanks
Merely for testing/research purposes I decided to try an open G729 codec
posted to this list a couple months ago. I tried it a couple times with
FreeSWITCH Version 1.0.trunk (11356M). And everything worked fine, while
starting FS I would get:
2009-04-20 13:29:59 [CONSOLE]
Hi,
I might be in a position (finally) to ask/suggest one of my voip providers
to use an alternative codec to G729. I wanted to know what would be the best
replacement for it.
Thanks again everybody for your time and info.
Regards,
Nicolas
___
I'm a native spanish speaker, I can help too!
Nicolás Brenner
On Tue, Apr 14, 2009 at 2:56 PM, Diego Viola diego.vi...@gmail.com wrote:
Hey guys,
If you need some Spanish help count with my help also.
Diego
On Tue, Apr 14, 2009 at 2:12 PM, Michael Collins m...@freeswitch.orgwrote:
Hi, I am not very familiar with FS internals, but I recently found this
new db engine called couchDB. Looks pretty interesting, and its main focus
is scalability.
Has anybody played with couchDB? does it make sense to replace sqlite with
couchDB in FS?
Here's a link to the project homepage:
-
geared more towards
storing dynamic datasets...rather ones that can be structured...like FS
calling data can.
But I might be wrong :)
your buddy.
--matt
On Mon, Apr 13, 2009 at 12:00 PM, Nicolas Brenner
nico...@medularis.comwrote:
Hi, I am not very familiar with FS internals, but I
Jacek,
I had a similar problem once. It actually depends on your sip gateway,
but I was able to solve the problem by setting the caller id, ie:
session1 = new Session();
session1.setCallerData(caller_id_name, 8280052500);
session1.setCallerData(caller_id_number, 8280052500);
Oops! Well, fortunately I don't use that voip provider anymore (nor the script).
Thanks Brian.
Nicolas
On Tue, Feb 3, 2009 at 2:25 PM, Brian West br...@freeswitch.org wrote:
YOU should NEVER use this method or call setCallerData at all you
should use the correct methods to override the
I would love to be a beta tester too! I haven't switched from Asterisk
for the same reason.
Cheers,
Nicolas
On Fri, Oct 10, 2008 at 4:38 PM, Michael Collins [EMAIL PROTECTED] wrote:
Can I at least be a beta tester or something? Please? I'm
desperate!!!
Dude, you're hired! :)
-MC
On Fri, Aug 1, 2008 at 9:51 AM, Anthony Minessale
[EMAIL PROTECTED] wrote:
...
A quad woodcrest 2.6ghz can do about 3000 simo media sessions with FS, the
same box can just make it to 400 when they are all G729 transcoding calls.
If they are bridged calls, that number goes in half, if we take
Brian,
Although I agree that it is not a good idea to split the community,
this wouldn't much split it as increase it. There's a lot of people
who don't understand english, but have the skills to use or learn
about FreeSwitch and even help in the development. Creating a latin
irc channel, could
I guess there's no interest in G729 at all... any good alternatives?
What do you guys use to save bandwith and keep a decent audio quality?
On Wed, Jun 4, 2008 at 7:34 AM, Nicolas Brenner [EMAIL PROTECTED] wrote:
Hi, I'd like to know what's needed to add support for G729, I know
there's
that has been resolved G729 will show up rather
quickly
K
From: Nicolas Brenner [EMAIL PROTECTED]
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Thu, 5 Jun 2008 08:42:50 -0400
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] g729
I guess there's no interest
Hi, I'd like to know what's needed to add support for G729, I know
there's a bounty, but I couldn't make sense out of what's posted on
the wiki. I'm really interested in this, as one of my current VoIP
providers restricts me to using only this codec, which limits me to
using Asterisk, hence I
:
you can do originate
{ignore_early_media=true,bypass_media=true}sofia/default/[EMAIL PROTECTED]
sofia/default/[EMAIL PROTECTED] inline
and hairpin 2 calls between the provider
On Thu, May 29, 2008 at 2:55 PM, Nicolas Brenner [EMAIL PROTECTED]
wrote:
Anthony and Ken (specially), thank you
if you decrease the calls per
second and increase the average call length. High Call Per Second Rates are
the bane of any switch
K
From: Nicolas Brenner [EMAIL PROTECTED]
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Thu, 29 May 2008 12:54:07 -0400
To: freeswitch-users
On Wed, Apr 9, 2008 at 7:36 AM, Brian West [EMAIL PROTECTED] wrote:
On Apr 9, 2008, at 1:17 AM, Nicolas Brenner wrote:
Hello everyone,
I'm having some trouble with FS :( apparently with mod_shout. I want
to play an mp3 file after answering a call so I compiled mod_shout
Thanks! It worked as advertised.
The only problem I have now, is my provider (I'm trying gafachi now),
I'm getting about one or two seconds delay on the audio, which is
pretty bad.
One other thing, it takes about 5 or 10 seconds to get a ring tone
after answering the first call, is there anyway
I made it work! the problem? my sip provider (Gizmo). I changed the
configuration to use voipdiscount (no comments), and the problem went
away.
By the way, I'm looking for good SIP providers. In the coming months
I'll need to handle a lot of load, and I'd also like good rates to
mobile phones in
Anyone has experience connecting Freeswitch with Net2Phone? Thanks!
On Tue, Mar 25, 2008 at 3:31 PM, Leonardo Alves [EMAIL PROTECTED] wrote:
the sip trunk in FS is the gateway.
Here is how you dial a gateway:
http://wiki.freeswitch.org/wiki/Sofia#Dial_out_of_a_gateway
And here is how to
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