Dear All,
I receiving the events in while loop by using recvEventTimed method in
ESL.pm. I have to flush that Event buffer after some particular time. How
can I do it?
Thanks,
Velusamy
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FreeSWITCH-users mailing list
Thanks Meftah for the help. So it looks like an SIP proxy is just to handle
the registration process, while a PBX offers services like ringing phones,
setting up calls, handling voice mail, etc.
--
View this message in context:
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read them in a timed loop of some small number of MS until you get a timeout
meaning you have flushed them all.
On Fri, Oct 30, 2009 at 1:57 AM, velusamy velu velu.techni...@gmail.comwrote:
Dear All,
I receiving the events in while loop by using recvEventTimed method
in ESL.pm. I have
It's strange... a tcpdump tells me that there is no DTMF from my provider when
using IVR, but when I call into a TN that goes directly into the Conference
App, I see DTMF from the provider.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
Hm, strange. I haven't seen that before. Can you pastebin your logs
at debug level?
On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote:
It's strange... a tcpdump tells me that there is no DTMF from my
provider when using IVR, but when I call into a TN that goes
directly into the
Hi all
I did a 'make update' to 15289 and I found the auto feature added to
mod_valet_parking so I figured I'd try it out. Everything works fine
except that the extension number for the parked call is reported on the
wrong leg of the call. It'd be good to sort this out as this is a
useful
Hey everyone! The weekly meeting will start in a few minutes. Here's the
agenda:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_30
Please feel free to add other subjects. If you have something to share with
the community that would be great. Also, if you have questions for the dev
team by all
You have to be doing it wrong then.
Can you show us your dialplan you should have two extensions one for
the lot range and one to attended transfer someone into the lot.
/b
On Oct 30, 2009, at 10:47 AM, Brian Stafford wrote:
Hi all
I did a 'make update' to 15289 and I found the auto
I am wondering why I cannot do as condition#2.
For Lua in dialplan, when I have the followings:
--WORKING--
(Condition#1)
.
.
session:execute(set,bypass_media=true)
session:execute(set,hangup_after_bridge=true)
session:execute(set,continue_on_fail=true)
.
.
Brian West wrote:
You have to be doing it wrong then.
Can you show us your dialplan you should have two extensions one for
the lot range and one to attended transfer someone into the lot.
/b
The relevant excerpt from the dialplan is
extension name=valet_unpark
condition
I have already set debug to 9, on both profiles.
Ivan
Den 29. okt. 2009 kl. 03:21 skrev Eliot Gable:
See that 200 OK that keeps coming in over and over and over and over
again? That's because they never received your ACK. If you can turn on
sofia loglevel to 9 and then watch where you send
Do you have a debug level message from switch_ivr_originate.c in your log?
Channel is already up, delaying proxy mode 'till both legs are answered.
Set bypass_media b4 each bridge. It is unsetting on you and setting
bypass_media_after_bridge because you already answered the channel running
the
fsctl loglevel debug
console loglevel debug
sofia profile internal siptrace on
sofia profile external siptrace on
sofia loglevel all 9
^
Then run your call, then do this:
sofia loglevel all 0
sofia profile external siptrace off
sofia profile internal siptrace off
fsctl
Hi Brian,
Thanks for the explanation. This seems to have solved the issue.
Robert.
-- Forwarded message --
From: Brian West br...@freeswitch.org
To: freeswitch-us...@lists.freeswitch.org
Date: Thu, 29 Oct 2009 13:31:26 -0500
Subject: Re: [Freeswitch-users] Gateway
Now i have as follows, but it's still the same result. By the way, I am
running: FreeSWITCH Version 1.0.4 (exported)
.
.
.
session:execute(set,hangup_after_bridge=true)
session:execute(set,continue_on_fail=true)
session:execute(set,originate_timeout=2)
session:execute(set,originate_retries=3)
Hi,
Which control panel are you using with FreeSwitch? WikiPbx, FreePbx or other?
Thanks,
Pedro Prado
_
Você sabia que com o Hotmail você tem espaço ilimitado para guardar seus
e-mails?
From watching the mailing list, I've seen:
95% are using straight XML combined with their home-cooked solutions based
on mod_xml_curl, which makes it pretty easy to write your own custom GUI for
freeswitch.
Out of the rest, the most are probably using pfsense, freepbx, and a small
handful of
2009/10/30 Pedro Prado pedropr...@msn.com
Hi,
Which control panel are you using with FreeSwitch? WikiPbx, FreePbx or
other?
These are all at various stages of production/usability:
Most of the people here just use emacs/vim/etc. and then use the built-in
webserver for simple stuff. (see
On Fri, Oct 30, 2009 at 5:06 AM, Fred-145 codecompl...@free.fr wrote:
Thanks Meftah for the help. So it looks like an SIP proxy is just to handle
the registration process, while a PBX offers services like ringing phones,
setting up calls, handling voice mail, etc.
Of course, as usual, the
Thanks for the link. But then, since OpenSIPs doesn't run on Windows, I'll
stick to FreeSwitch :)
--
View this message in context:
http://old.nabble.com/PBX-vs.-SIP-proxy--tp25920788p26136363.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.
Yes, now I got a more detailed trace. Thank you for helping me with
this.
A new pastebin at http://pastebin.freeswitch.org/10905
Ivan
Den 30. okt. 2009 kl. 18:30 skrev Eliot Gable:
fsctl loglevel debug
console loglevel debug
sofia profile internal siptrace on
sofia profile external
thanks for the heads up! I'll ask someone in the know to fill that out.
-MC
On Fri, Oct 30, 2009 at 1:28 PM, Diego Viola diego.vi...@gmail.com wrote:
This needs to be converted to documentation form also.
http://wiki.freeswitch.org/wiki/Proxy_Media#How_to_detect_when_it_should_be_used
On
No problem, when I have some time I need to merge these two pages also.
http://wiki.freeswitch.org/wiki/Multi-tenant
http://wiki.freeswitch.org/wiki/Multiple_Companies
=D
Diego
On Fri, Oct 30, 2009 at 9:05 PM, Michael Collins m...@freeswitch.org wrote:
thanks for the heads up! I'll ask
does anybody know how does it work and how to use it in a dialplan?
freeswi...@nemesis
freeswi...@nemesis
freeswi...@nemesis load mod_t38gateway
API CALL [load(mod_t38gateway)] output:
+OK
2009-10-30 22:44:38.204268 [NOTICE] mod_t38gateway.c:147 T.38 gateway
enabled
2009-10-30 22:44:38.204268
Hi everybody,
I'd like some help with this situation that is 'haunting' me :-)
My scenario is as follows:
inbound-bypass-media is set in the profile because we dont want FS handling the
media.
1. A calls B
2. FS sends to B the A's SDP
3. B answers
4. FS sends to A the B's SDP
5. Media going
Another one to convert:
http://wiki.freeswitch.org/wiki/User:Agx#CAMP_ON
On Fri, Oct 30, 2009 at 9:36 PM, Diego Viola diego.vi...@gmail.com wrote:
No problem, when I have some time I need to merge these two pages also.
http://wiki.freeswitch.org/wiki/Multi-tenant
actually that's a user's personal page so you don't have to worry about that
one. :)
-MC
On Fri, Oct 30, 2009 at 3:09 PM, Diego Viola diego.vi...@gmail.com wrote:
Another one to convert:
http://wiki.freeswitch.org/wiki/User:Agx#CAMP_ON
On Fri, Oct 30, 2009 at 9:36 PM, Diego Viola
First you forgot to mention what SVN rev you're on...
/b
On Oct 30, 2009, at 5:07 PM, Humberto Quintana wrote:
Hi everybody,
I'd like some help with this situation that is 'haunting' me :-)
My scenario is as follows:
inbound-bypass-media is set in the profile because we dont want FS
profile options
param name=media-option value=resume-media-on-hold/
param name=media-option value=bypass-media-after-att-xfer/
On Fri, Oct 30, 2009 at 5:07 PM, Humberto Quintana hjqlo...@hotmail.comwrote:
Hi everybody,
I'd like some help with this situation that is 'haunting' me :-)
My
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