wrote:
try
apiExecute(uuid_media, off + session.uuid);
On Wed, Jul 1, 2009 at 3:22 PM, Phillip Jones pjinthe...@gmail.comwrote:
Hi there,
I was wondering whether it is possible to have FreeSwitch go into
bypass_media mode on demand?
For instance, leg a bridges to leg b - leg b
are doing.
you could set the variable bypass_media_after_bridge=true on the a leg
before you call the b leg and use the group_confirm feature to get the
caller
to press the key.
On Thu, Jul 2, 2009 at 10:41 AM, Phillip Jones pjinthe...@gmail.comwrote:
Thanks for that.
That seems
script
On Thu, Jul 2, 2009 at 11:53 AM, Phillip Jones pjinthe...@gmail.comwrote:
Thanks for responding and for your help.
The xml and confirm.js are attached below. Basically trying to
bypass_media after the leg B presses 1 to accept the call. I tried,
using bypass_media_after_bridge=true
(x86)\Freeswitch\sounds\en\us\callie\digits\8000
This is a clean install of FreeSWITCH - so I am wondering why I needed to do
this, what have not configured correctly?
As you can see I am using windows with a resent build (3 days) from svn.
Any help appreciated.
Thanks
Phillip Jones
dir for digits would
be specified?
On Thu, Jul 9, 2009 at 12:38 PM, Michael Collins m...@freeswitch.org wrote:
Is the call perhaps at 16kHz and it's looking for non-installed 16kHz sound
files?
-MC
On Thu, Jul 9, 2009 at 9:06 AM, Phillip Jones pjinthe...@gmail.comwrote:
Hi there,
I have
Ok - forget this one - I did a fresh install from the pre-compiled windows
binary/msi - referenced on the wiki - and every thing is working as it
should be.
Thanks
On Thu, Jul 9, 2009 at 2:46 PM, Phillip Jones pjinthe...@gmail.com wrote:
Thanks for the response.
I don't think so - the trace
)\FreeSWITCH\mod and NOT C:\Program Files (x86)\FreeSWITCH\mod\managed
thus making C:\Program Files (x86)\FreeSWITCH\mod\managed redundant
Anyone else seen this behavior?
Thanks!
Phillip Jones
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should put the references in managed directory.
Check this link
http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-April/002232.html ,
may be your case
Diego
--- On *Fri, 7/10/09, Phillip Jones pjinthe...@gmail.com* wrote:
From: Phillip Jones pjinthe...@gmail.com
Subject
-users-boun...@lists.freeswitch.org [mailto:
freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Phillip
Jones
*Sent:* Friday, July 10, 2009 11:46 AM
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* [Freeswitch-users] managed_mod directories
Hi there,
Using windows
on mod/managed although I changed loadfile by loadfrom on loader.cs.
Diego
--- On *Fri, 7/10/09, Phillip Jones pjinthe...@gmail.com* wrote:
From: Phillip Jones pjinthe...@gmail.com
Subject: Re: [Freeswitch-users] managed_mod directories
To: freeswitch-users@lists.freeswitch.org
Date: Friday
classes - you should place your code
in
your own dll.
- Jeff
Phillip Jones-2 wrote:
Hi,
If I place the DLL in mod\managed I get the following error:
[err] mod_managed.cpp:287 Assembly::LoadFrom failed:
system.IO.FileNotFoundException: Could not load file or assembly
'file
until the bridge is completed. I need it
to reset when leg b is answered.
I tried resetting the leg_timeout in the confirm script after leg b is
answered. I also tried using leg_progress_timeout. Neither seemed to work.
Any help or suggestions would be welcome.
Phillip Jones
the call is answered. Not when
the called party presses 1 and the bridge is complete.
I am new all this so I will work out how to use the pastebin etc.
Thanks for your help.
Phillip Jones
On Tue, Jul 14, 2009 at 12:16 PM, Michael Collins m...@freeswitch.orgwrote:
On Tue, Jul 14, 2009 at 8:35
seconds before I sent you the email.
On Tue, Jul 14, 2009 at 1:14 PM, Phillip Jones pjinthe...@gmail.comwrote:
Thanks for your response.
That does not seem to work. Here is my code:
if(Session.Ready())
{
Session.Execute(set, ignore_early_media=true);
Session.Execute(set
Hi there,
I am sure I am missing something. Can someone point out where to signup for
a username / password to pastebin.freeswitch.org.
I am pulling my hair out and feel kinda stupid asking this.
Thanks
Phillip Jones
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Thank you!
Not that obvious in IE actually - I should stick to FF.
On Tue, Jul 14, 2009 at 5:43 PM, Michael Collins m...@freeswitch.org wrote:
On Tue, Jul 14, 2009 at 2:27 PM, Phillip Jones pjinthe...@gmail.comwrote:
Hi there,
I am sure I am missing something. Can someone point out
=MyFSProjects.InboundCall/
confirm.js is at: http://pastebin.freeswitch.org/9713
Thanks again for your help on this.
On Tue, Jul 14, 2009 at 2:35 PM, Phillip Jones pjinthe...@gmail.com wrote:
Ah - SVN Trunk - thought you meant DID trunk!!! My bad.
Sorry - understand now! Will recompile
much appreciated.
Phillip Jones
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:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Phillip
Jones
*Sent:* Thursday, July 16, 2009 3:08 PM
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* [Freeswitch-users] .NET demo / hangupHook()
Hi there,
I am
Hi there,
Very simple scenario:
Session.DtmfReceivedFunction = (d, t) =
{
Log.WriteLine(LogLevel.Info, Received {0} for {1}., d, t);
CollectedDigits = d.ToString().Trim();
return ;
};
Session.flushDigits();
Session.StreamFile(VoicemailPromptsDirectory + abigfile.wav, 0);
Question
;
};
Session.SayPhrase(msgcount, 187346, en);
Any ideas? I am sure I must be missing something simple.
Thanks a lot.
Phillip Jones
On Thu, Jul 23, 2009 at 2:40 PM, Michael Collins m...@freeswitch.org wrote:
I think you might want to check out phrase macros...
http://wiki.freeswitch.org/wiki
.
Thanks a lot.
Phillip Jones
On Thu, Jul 23, 2009 at 2:40 PM, Michael Collins m...@freeswitch.org
wrote:
I think you might want to check out phrase macros...
http://wiki.freeswitch.org/wiki/Speech_Phrase_Management
-MC
On Thu, Jul
a prime example is a b-leg whisper where you are still playing
ringing to the a-party while
the b-party answers and hears a message just before connection to the
a-party.
You should be able to do this particular function using group_confirm. See
this page in the wiki.
Mike,
I am not familiar with Asterisk so I am not 100% sure this is what you are
looking for, but check out
http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation
It may contain what you need
Phillip Jones
On Mon, Aug 3, 2009 at 3:23 PM, Michael Frager m
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 12:27:44.229091 [NOTICE] sofia.c:322
This must be a fairly comment requirement so any ideas on what I might
be missing would be very welcome.
Thanks
Phillip Jones
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Hi there,
Not sure whether this helps but test this without set bypass_media. In
my setup I have noticed the leg A session ends when bypass_media is
true. Call/bridge continue successfully.
Phillip Jones
On Thu, Aug 6, 2009 at 1:28 PM, Benedikt
Fraunhoferfraunhofer.lists.freeswitch
be missing would be very welcome.
Thanks
Phillip Jones
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Hi there,
Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re
INVITE) and only pass back the media to the network, or pass back
signaling also (SIP REFER)?
I know several suppliers who support SIP re INVITE but none that
support SIP REFER.
Check out
Thanks very much for that - very help.
Why would loopback be considered abuse? What would be the downsides
of doing this?
On Sat, Aug 8, 2009 at 7:18 AM, Rupa Schomakerr...@rupa.com wrote:
On Sat, Aug 8, 2009 at 9:16 AM, Rupa Schomaker r...@rupa.com wrote:
action application=bridge
.
-Vladimir Rodionov
On Sat, Aug 8, 2009 at 10:40 AM, Phillip Jones pjinthe...@gmail.com
wrote:
Hi there,
Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re
INVITE) and only pass back the media to the network, or pass back
signaling also (SIP REFER)?
I know several suppliers who
not careful with them.
Mike
On Aug 8, 2009, at 1:44 PM, Phillip Jones wrote:
Thanks very much for that - very help.
Why would loopback be considered abuse? What would be the downsides
of doing this?
On Sat, Aug 8, 2009 at 7:18 AM, Rupa Schomakerr...@rupa.com wrote:
On Sat, Aug 8, 2009 at 9:16
of what
that means.
Mike
On Aug 8, 2009, at 4:24 PM, Phillip Jones wrote:
Mike/Rupa ,
Thanks for your help on this. So I am correct that summarizing that
FreeSWITCH does not really support fail over and multiple call
destinations because the same mechanism is used to achieve both
and sending call directly through
the gateway to which the loopback points.
e.g. http://pastebin.freeswitch.org/9948
Has anyone else encountered this? Am I missing something?
Any help would be much appreciated!
Thanks
Phillip Jones
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anyone else encountered this? Am I missing something?
Any help would be much appreciated!
Thanks
Phillip Jones
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, 2009, at 9:46 AM, Phillip Jones wrote:
Thanks for the reply Brian.
I went through this fairly carefully with Mike Jerris - please see
http://www.nabble.com/Calling-Multiple-Destinations-with-Failover-td24877157.html
I am very open to any other mechanism that allows the calling multiple
by the way - but point taken.
Any other ideas on how to achieve multiple destinations with carrier
failover would great!
Thanks again.
On Tue, Aug 11, 2009 at 2:07 PM, Michael Collinsm...@freeswitch.org wrote:
On Tue, Aug 11, 2009 at 11:08 AM, Phillip Jones pjinthe...@gmail.com
wrote:
Can
Hi there,
I the 'does not work' example your media stream is back up at the
carrier as the called party hangs up.
You would have to cause another SIP REINVITE and re capture the media.
See http://wiki.freeswitch.org/wiki/Bypass_Media and the How to
disable/enable it on the fly? section as a
.
Again - thank you all for your help. You do a great job of getting
getting your heads around all this stuff day in day out.
Phillip Jones
On Tue, Aug 11, 2009 at 3:19 PM, Michael Collinsm...@freeswitch.org wrote:
On Tue, Aug 11, 2009 at 1:43 PM, Phillip Jones pjinthe...@gmail.com wrote:
Hi
loopback working with
bypass_media.
In the meantime I will also start looking to build a custom bridging
app. As I said though - not a road I wanted to go down.
Thanks for your help!
Phillip Jones
On Tue, Aug 11, 2009 at 6:13 PM, Michael Giagnocavom...@giagnocavo.net wrote:
It's also simple enough
!
Phillip Jones
On Tue, Aug 11, 2009 at 6:13 PM, Michael Giagnocavom...@giagnocavo.net
wrote:
It's also simple enough to write a plugin in one of the scripting languages
to add an app to do exactly what you want...
-Original Message-
From: freeswitch-users-boun
into freeswitch - no more starting
and stopping freeswitch!
Also appreciate the f# example.
Thanks again.
Phillip Jones
On Wed, Jul 29, 2009 at 8:04 PM, Michael Giagnocavom...@giagnocavo.net wrote:
Which directory ends with ; ?
I’m not following – if you want email me off list and we can work
Rupa / all,
Just a quick follow up to this.
This is appears to a timing issue. If I try and do a uuid_media off +
uuid in api_after_bridge it fails with CHAN_NOT_IMPLEMENTED
and the call is dropped.
If appears to be trying to do a SIP reinvite on the loopback channel
which is of course just
(514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 13-Aug-09, at 6:59 PM, Phillip Jones wrote:
Rupa / all,
Just a quick follow up to this.
This is appears to a timing issue. If I try and do a uuid_media off +
uuid in api_after_bridge it fails
that supporting groupings of
carriers is would be the most elegant solution.
Let me know if I am just talking rubbish re the api_after_bridge event.
Thanks!
Phillip Jones
On Fri, Aug 14, 2009 at 2:33 AM, Rupa Schomakerr...@rupa.com wrote:
On Thu, Aug 13, 2009 at 6:54 PM, Mathieu Renemrene_li
be great.
Thanks for every bodies help on this.
Phillip Jones
On Fri, Aug 14, 2009 at 3:17 PM, Phillip Jonespjinthe...@gmail.com wrote:
Hi Rupa,
What about my suggestion above introduce a api_after_bridge event
that fires when the switch_ivr_uuid_bridge() bridges to the two sofia
channels
Take a look at http://jira.freeswitch.org/browse/FSCORE-422. This a
feature request I submitted. This problem it solves is different - but
the solution is the same. Perhaps you add your take to the comments
there.
On Tue, Aug 25, 2009 at 10:06 AM, Carlos S. Antunesc...@nowthor.com wrote:
Max,
Hi there,
I know there are other ways of doing this. I am just trying get to
know have fun with the FreeSWITCH API.
I am using originate and park a call:
fsApi.Execute(originate,
string.Format([origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
park,
app::arg
there are variables you can set that begin group_confirm_* that you can set
to do this all without even using a script.
http://wiki.freeswitch.org/wiki/Channel_Variables#Answer_confirmation_variables
On Fri, Aug 28, 2009 at 3:53 PM, Phillip Jones pjinthe...@gmail.comwrote
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 nick.lember...@lkfd.netwrote:
thanks
Phillip Jones
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/ managed equivalent of:
switch_event_bind(const char *id, switch_event_types_t event, const char
*subclass_name, switch_event_callback_t callback, void *user_data);
Thank you!
Phillip Jones
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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 there,
Can anyone give any insight to this following message:
2009-09-02 15:21:44.665608 [CRIT
.
-Michael
*From:* freeswitch-users-boun...@lists.freeswitch.org [
mailto:freeswitch-users-boun...@lists.freeswitch.orgfreeswitch-users-boun...@lists.freeswitch.org
] *On Behalf Of *Phillip Jones
*Sent:* Thursday, September 03, 2009 8:27 AM
*To:* freeswitch-users@lists.freeswitch.org
Phillip Jones-2 wrote:
Hi there,
mod_managed exposes EventReceivedFunction such that:
Session.EventReceivedFunction = (e) =
{
Log.WriteLine(LogLevel.Alert, Received Event {0}, e.ToString());
return ;
};
should trap all events to which i subscribe.
But how do
Hi there:
The wiki states:
uuid_exists
Checks whether a given UUID exists.
Usage: uuid_exists uuid
However when I call via an API call I get:
INVALID COMMAND!
I also don't see it in MOD_COMMAND.C
Has it gone?
Thanks
Phillip Jones
Surely this is because you are calling ev.serialized_string and ev is null
if there is not event waiting?
i.e. con.pop(0); returns null if no event is waiting.
Therefore
if(ev != null)
Log.WriteLine(LogLevel.Notice, Event: + ev.serialized_string);
would work, alternatively calling
it's a rite of passage :)
On Mon, Sep 14, 2009 at 11:29 AM, Jerry Richards jerry.richa...@teotech.com
wrote:
Aha... I have been notified that I failed the test. The username/password
is given in the authentication pop-up itself. My bad...
-Original Message-
From: Jerry
My FS based ACD allows agents to log-in to multiple queues at once
What is the difference between those queues? Does each q require that agents
have a different skill? ie Agent A has skills x,y,z and queue 1,2,3 have
calls about x,y,z - there agent A logins to queue 1,2,3.
2009/9/17 Paweł
see here: *http://en.wikipedia.org/wiki/Skills_based_routing*
On Thu, Sep 17, 2009 at 4:42 PM, Phillip Jones pjinthe...@gmail.com wrote:
My FS based ACD allows agents to log-in to multiple queues at once
What is the difference between those queues? Does each q require that
agents have
session.dispose();
???
On Thu, Sep 17, 2009 at 5:20 PM, Alberto Escudero aep.li...@it46.se wrote:
We are trying to create a callback application in Javascript. We get the
callerid from the unanswered call and after destroying the session, we
initiate a callback to the user to conenct it to a
I would be interested in this too
Concerning mod_fifo - can you restrict an agent to a slot. So lets say DNIS
A is for product A and DNIS B product B - some agents know both - some know
just 1 product - would that be possible?
On Thu, Sep 17, 2009 at 3:17 PM, Anthony Minessale
Take a look at http://wiki.freeswitch.org/wiki/Mod_xml_curl to get started.
On Fri, Nov 13, 2009 at 4:59 PM, Jerry Richards
jerry.richa...@teotech.comwrote:
Is there a way to access configuration information from a database (e.g.
SQL) rather than from the XML files?
Best Regards,
Jerry
Hi there,
I have created a simple conference that works great. The only problem is,
when a participant press # it exits the call. So when a user enters a
conference with a PIN, and by habit they enter 12345 followed by pound, it
puts them in and then straight out.
So I edited conference.conf.xml
, 2009 at 2:35 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
issue
console loglevel debug
from the cli
then try again and see if there is any hint
On Mon, Nov 23, 2009 at 1:24 PM, Phillip Jones pjinthe...@gmail.comwrote:
Hi there,
I have created a simple conference
=caller-controls value=default/
Do you want me to test this on the latest trunk or is this as expected?
Phil
On Mon, Nov 23, 2009 at 3:12 PM, Michael Collins m...@freeswitch.org wrote:
On Mon, Nov 23, 2009 at 11:24 AM, Phillip Jones pjinthe...@gmail.comwrote:
Hi there,
I have created
...@freeswitch.org wrote:
On Mon, Nov 23, 2009 at 12:17 PM, Phillip Jones pjinthe...@gmail.comwrote:
Thanks for replying.
Well in the log I see:
2009-11-23 15:13:22.015625 [DEBUG] switch_rtp.c:2282 RTP RECV DTMF #:760
2009-11-23 15:13:22.062500 [DEBUG] mod_conference.c:2379 Channel leaving
hi there,
I have set up some cisco 7960 up with fs. They work fine - but the only way
I can keep them registered is to set the timer_register_expires in the
Cisco cfg file to something really short like 10s.
Does anyone know the default register timeout for fs? And where I might
change this in
Thanks for this goodness. I am sure to use it so it is appreciated.
On Mon, Nov 30, 2009 at 2:51 PM, Andrew Thompson and...@hijacked.us wrote:
Tony committed my patch for doing 'week of month' conditions in the XML
dialplan along with some holiday routing examples to the default
dialplan. Now
Hi there,
It it possible do something like:
extension name=rtp
condition field=destination_number expression=^2127776252$
action application=answer/
action application=playback data=rtp://192.563.41.246:27378/
/condition
/extension
Basically I have need to connect to incoming calls
Hi All,
Every so often you have to ask a question - where you know so little - it's
hard to even now where to start. This is one of the times. I am not
expecting an full answer here, just a gentle nudge in right direction to get
me started.
What I have is a propriety IP based conference system -
to your callers, no?
After that, then I would imagine you just need to do SDP rewriting when a
call hits FreeSWITCH.
-Michael
*From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Phillip
Jones
*Sent:* Friday
Also check out this great write up:
http://wiki.freeswitch.org/wiki/Multi_home_tutorial
This is pfSense specific.
On Sat, Dec 5, 2009 at 10:22 AM, ram talk2...@gmail.com wrote:
On Sat, Dec 5, 2009 at 3:16 PM, mailinglist mailingl...@fribert.dkwrote:
Has anybody done this?
I'm
Hi - sorry to go off topic - but we are looking for Voip supplier with SMS
capability. Would you mind telling me which Voip supplier you use?
On Thu, Dec 10, 2009 at 11:10 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
My voip provider provides a SOAP interface to be able to
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