seven ha scritto:
oh, thank you Antonio. I think it would be better to collect more
ideas before open a bounty. And I more interested in playing(including
patching the code) with that than use the function.
I was working on other stuff yesterday and just looked at the wiki:
- it seems
I looked at FS code directly and had an improvement about the problem.
Topic 1: presence can be changed and working
The wiki page:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_presence is totally
wrong about the presence
application, indeed when i first asked on IRC some weeks ago i
Hi,
It should be easy to modify mod_fifo to include this functionality.
Correct me if I'm wrong:
For call back agents at least, when X calls are in the the queue,
Freeswitch tries to search for up to X agents in database. This
algorithm is much more optimized than Asterisk, as Asterisk will take
EdPimentl wrote:
Here is a list of the resources posted on this thread
After giving it more thoughts, I got to the conclusion that I'd rather a
stand-alone miniPC that can take a PCI card with a riser, instead of a
really tiny box that relies on an external box to connect to a PSTN line.
By IDE, I'm assuming you mean the all too familiar 40 pin PATA.
Yes, as a matter of fact:
http://www.newegg.com/Product/ProductList.aspx?Submit=ENEN=2003240636+1421530855Configurator=Subcategory=636description=Ntk=SpeTabStoreType=srchInDesc=
Do some research since they're not all created equal.
On Wed, Apr 29, 2009 at 2:01 AM, Antonio Gallo ga...@mctelefonia.comwrote:
I looked at FS code directly and had an improvement about the problem.
Topic 1: presence can be changed and working
The wiki page:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_presence is totally
wrong
It depends, not every phone does presence right and they all have
different levels of support.
The 2 you are using are very different from how the majority of them
work (snom, polycom, linksys etc)
Did you try reversing the role of A and B
I will pickup a snom320 and a gxp 20xx and will try
Hello.
Can I use custom variables in the condition field?
For example if i set my custom var ${forward_all} and continue search extensions
condition field=${forward_all} .../ not work even if the condition is met
and call does not go in the uplink.
Or, in the condition I can use only variables
On Wed, 2009-04-29 at 04:47 -0700, Fred-145 wrote:
EdPimentl wrote:
Here is a list of the resources posted on this thread
After giving it more thoughts, I got to the conclusion that I'd rather a
stand-alone miniPC that can take a PCI card with a riser, instead of a
really tiny box that
OK let me explain this for the 100th time. :)
The dialplan isn't executed line by line. The dialplan is just a
compiled list of instructions to execute but its execution doesn't
take place until the session enters the execute state. So your set
forward_all= hasn't happened yet. BUT
There should be several archive threads on this including:
http://n2.nabble.com/Using-Variables-in-Dialplans-tt2678222.html#a2678222
Mike
On Apr 29, 2009, at 10:30 AM, Alex Gusak wrote:
Hello.
Can I use custom variables in the condition field?
For example if i set my custom var
All of a sudden I'm getting this startup error when I start FreeSwitch:
C:\DVLP\FreeSwitchfreeswitch
Error including
C:\DVLP\FreeSwitch\conf\autoload_configs\..\sip_profiles\internal/*.xml
(Invalid argument)
Cannot Initialize [[error near line 2423]: unexpected closing tag
/context]
However,
The first is an error that is unrelated to the second error.
Check out freeswitch.xml.fsxml line 2423 you'll have an extra /
context line there.
/b
On Apr 29, 2009, at 10:21 AM, Gerry Hull wrote:
All of a sudden I'm getting this startup error when I start
FreeSwitch:
At least your dialplan should have a tag named context. See default dialplan !
Original Messageprocessed by David.InfoCenter
Subject: [Freeswitch-users] Very confusing startup error (29-Apr-2009 17:26)
From:Gerry Hull ge...@pstn2.net
To: g...@exram.de
All of a sudden I'm
Thanks Guys!
I could not find my problem -- but you pointed me in the correct
direction. I had a mismatched tag in my public.xml in the dialpan.
So, is freeswitch.xml.fsxml a logged representation of the complete config
file in memory?
On Wed, Apr 29, 2009 at 11:36 AM, Guido Kuth
Hi guys ..
I need your help please...
I am trying to setup an FS box. It has to be like a 3 way thing since i
reside in one network - my FS machine resides on another network - and my
provider (gateway) resides on another network.
I am going to try to be specific as much as i can.
I did a Quick
ok i did some test today using the yesterday's trunk with a gxp2010 and
a snom360 both with 2 LEDS monitoring each other and themselves.
Configuration:
gxp2010 user: 1000 led1: 1000 led2: 1001
snom360 user: 1001 led1: 1000 led2: 1001
Problem with both phones:
- when a phone reboot and
Now you need to open up the sofia profile in sip_profile/internal.xml
and apply the test1 acl instead of the domains acl.
/b
On Apr 29, 2009, at 11:12 AM, Edward Q. wrote:
freeswi...@internal 2009-04-29 11:46:05 [DEBUG] sofia.c:4242
sofia_handle_sip_i_invite() IP 75.74.xxx.xxx Rejected by
On Wed, Apr 29, 2009 at 8:54 AM, Gerry Hull ge...@pstn2.net wrote:
Thanks Guys!
I could not find my problem -- but you pointed me in the correct
direction. I had a mismatched tag in my public.xml in the dialpan.
So, is freeswitch.xml.fsxml a logged representation of the complete config
After 6 months of discussions with Attractel, today we finally got a new
version of Zoiper Bizz, which works with TLS and SRTP (previous versions
only supported TLS).
I have added the info, how to set it up, in the wiki
http://wiki.freeswitch.org/wiki/Interop_List#Zoiper_Bizz_2.10_and_TLS.2FSRTP
I have a problem I am trying to solve for several days now. I have FS 1.3.0
installed. I have the default configuration except that I have edited
event_socket.conf to match my configuration. I have two computers with x-Lite
SIP phone 1000 and 1001. Both started and registered. I call in from
If you subscribe to the event you will receive one on every DTMF press
if FreeSWITCH gets it... if you happen to be getting them via inband
you won't receive an event unless you enable the inband detection app.
http://wiki.freeswitch.org/wiki/Event_list#DTMF
set the async flag on the socket app call that triggers your ESL connection
On Wed, Apr 29, 2009 at 12:21 PM, Guido Kuth g...@exram.de wrote:
I have a problem I am trying to solve for several days now. I have FS
1.3.0 installed. I have the default configuration except that I have edited
First thanks for your reply.
I have subscribed to all Events, so this can't be the mistake. I sent
start_dtmf app to FreeSwitch in caller channel and the wiki says that you have
to do this on sip channels to enable inband dtmf. I checked sofia.conf and I
have found that param dtmf-type is
Hello Anthony,
sorry, but I forgot to tell you that I have an inbound ESL connection not an
outbound one. So I connect to FS and then wait for Events. I know that I can
set async flag in outbound socket, but is this also possible for inbound
socket, and when, is it the same as in outbound
Well the best option is to NOT use inband at all if possible. And use
RFC2833 which eyebeam/xlite support as do most providers out there...
You do not HAVE to start_dtmf on sip channels unless they only send
the DTMF inband.
set the dtmf-type back to rfc2833 and restart FS.
/b
On Apr
I use FS 1.0.3 with Grandstream HT386 and X-Lite soft phones, FS binds
to Mysql DB for SIP registrations, presence etc.
I noticed that after some time probably 30 min. phones which have been
registered but without making calls become unreachable. Meaning that any
call to such extension gets
Thank you again Brian. The reason why I want to test with inband dtmf is that
in the real environment FS will be behind a conventional ISDN PBX which will
work as a gateway to the ISDN Network. So I do not know if the PBX will do
something like a translation between DTMF Tones to rfc and
Do the phones and FS have a firewall between them? If so, sounds like
the pin hole in the fw is being closed. Alot only stay open for 4 mins
Regards,
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf
They do, but all necessary ports for FS are open. If that is fw issue,
are there ways to fight with it?
Nik Middleton wrote:
Do the phones and FS have a firewall between them? If so, sounds like
the pin hole in the fw is being closed. Alot only stay open for 4 mins
Regards,
-Original
Don't know where the setting is in FS, but force them to register every
120 seconds and see if that helps
Regards,
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
paul.degt
Sent: 29 April 2009
None that I know of, but it should be fairly simple to create FOP for FS
with the event socket.
2009/4/29 Antonio Gallo ga...@mctelefonia.com
ok i did some test today using the yesterday's trunk with a gxp2010 and
a snom360 both with 2 LEDS monitoring each other and themselves.
On Wed, Apr 29, 2009 at 2:13 PM, Mikael Bjerkeland mik...@bjerkeland.comwrote:
None that I know of, but it should be fairly simple to create FOP for FS
with the event socket.
The fairly simple part is actually doing it. The really hard part is
finding the time/energy/inclination to do it...
edit autoload_configs/sofia.conf.xml
in global_settings
add
param name=debug-presence value=2/
then you will see all the sql stmts etc and you can debug your issue
On Wed, Apr 29, 2009 at 11:18 AM, Antonio Gallo ga...@mctelefonia.comwrote:
ok i did some test today using the yesterday's
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