Lloyd,
I used this sometime ago to setup ours...
http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR
--
Jarrod Lash, jar...@fed-com.com
Federated Communications, LLC.
www.fed-com.com
Office: +1-412-357-2127
Mobile: +1-412-999-0049
Fax: +1-412-545-8368
On Mon, Dec 14, 2009
Hi All,
I am trying to setup a conference room with pin number authentication. I
could not find any wiki documents. If some one help me that would be
helpful.
Thank you in advance.
Thanks
Lloyd
___
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http://wiki.freeswitch.org/wiki/Mod_conference
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 14-Dec-09, at 10:06 PM, Aloysius Thevarajah Lloyd wrote:
Hi All,
I am trying to setup a conference room with pin number
On Thu, Oct 15, 2009 at 3:44 AM, god.nirvana god.nirv...@gmail.com wrote:
hi all
how can i get the digits when users in the conference??
and,in conference.conf.xml
control action=mute digits=0/ the action will set another
value?e.g:transfer?
thanks
I'm not sure I
?
___
Thanks, Nikita
_
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Joao
Mesquita
Sent: Wednesday, October 14, 2009 7:38 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] conference
hi all
how can i get the digits when users in the conference??
and,in conference.conf.xml
control action=mute digits=0/ the action will set another
value?e.g:transfer?
thanks
2009-10-15
god.nirvana
___
FreeSWITCH-users
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Joao
Mesquita
Sent: Wednesday, October 14, 2009 7:38 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] conference call
Look at eavesdrop on the wiki.
JM
2009/10/14 Nikita Belov nbe...@abisoft.spb.ru
HI all,
I want to configure FS to make special conference call between three users
(A, B, C). In this conference C will hear A and B, but A will hear only B.
Can I make it using FS API commands? Does
Hi folks!
Suddenly I found this
http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-February/002015.htmltopic
and that explains a lot.
From there I see that sofia sends refresher messages for NATed client in
order to check if it still alive.
It means I have problems in my client. Sorry for
I am still experiencing problem with lost media in conference on a client
behind NAT.
This is what I've done - disabled VAD on a NATed client and asked my friend
to produce lots of animal sounds in order to keep channel busy. But at the
end of minute sounds of wild nature disapeared again. We
And here is a short piece of log from the server side:
...
nua(): refersh session after 62 seconds (in [55..65])...
send INVITE ...
rcv OK...
send ACK...
rcv BYE...
I see now that sdp for natted client has additional lines in OK response
compared to client with public ip.
Session-Expires:
I am a bit confused with what's going on in a following scenario.
I have a public FS server with a public conference, that clients are
connecting to with my softphone. All of this softphones have STUN option
enabled and working, effectively resolving client's public IP address. They
also have ICE
RobertT siniy...@gmail.com wrote:
Where is the problem? Is it NAT, closing RTP port after some silence period
from client?
It could be a time-out, i.e., the nat router isn't keeping the port
translation alive.
I don't like nat at all. As more people migrate to IPv6 the problem will
gradually
Are there ways to escape this timeouts exchanging RTP with FS? Why didn't
waste flag help? Maybe I should flood channel in both directions? Will CNG
on a client side be a good descision? =)
___
FreeSWITCH-users mailing list
ты все еще наблюдаешь эту проблему?
я думал она уже решена...
эни вей, я уже приехал и сделаю скоро воторой IP нам для собственного
STUN-сервера.
--
Best regards,
Dmitry Kadantsev
http://www.doxwox.com - Best web meeting and online collaboration tool.
On Tue, Sep 29, 2009 at 10:32 AM, RobertT
в том то все и дело что с тобой мы эту проблему вроде как решили, а у Юры ее
никогда не было. и тут на тебе...
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Most likely the client NAT is cutting off the translation due to no
traffic. This could be because the client is not sending any traffic,
regardless of settings you set on FreeSWITCH. Try disabling all vad
and dtx on your soft phone to see if this helps. Also, your email
seems to
Hi all,
I've written a C# module for FS that creates conference dialplans on the
fly. From my limited understanding the easiest way to do this is by writing
XML to the directory: conf/dialplan/default/ - so this is the approach I've
taken.
After some suggestions in IRC to remove errant
Of Jason
White
Sent: Wednesday, 23 September 2009 4:47 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Conference with pin dialplan
Craig Presti cr...@vastpark.com wrote:
I've written a C# module for FS that creates conference dialplans on
the fly. From my limited
you must have some tdm equipment somewhere that is decoding the dtmf tones
and passing them w/o removing them from the audio stream.
On Tue, Sep 8, 2009 at 12:26 PM, Bradley Brashier bjbrash...@gmail.comwrote:
I have a FreeSWITCH conference with a list of DTMFs, some of which are
handled
Good thought. I'll look into that.
BB
On Tue, Sep 8, 2009 at 11:48 AM, Anthony
Minessaleanthony.miness...@gmail.com wrote:
you must have some tdm equipment somewhere that is decoding the dtmf tones
and passing them w/o removing them from the audio stream.
On Tue, Sep 8, 2009 at 12:26 PM,
Its your gateway provider not squelching the DTMF I suspect.
/b
On Sep 8, 2009, at 12:26 PM, Bradley Brashier wrote:
The issue is that when a command is pressed on one phone in the
conference, all users hear the tones of the first key pressed. My
expectation is that no other users should
I have a FreeSWITCH conference with a list of DTMFs, some of which are
handled through the event socket (like mute-all), some of which are
handled by FreeSWITCH itself (like mute-self). There are a number of
commands available and all of them are 2 digits in length.
The issue is that when a
thank you
but we defined the conference with
param name=caller-controls value=none/
so no keys should be available
br
On 2009-09-02 17:02, Andy Spitzer wrote:
Woof!
On Tue, 01 Sep 2009 18:52:01 -0400, Anthony
Minessaleanthony.miness...@gmail.com wrote:
there is no chance that you
hello
we have got a little problem with the conference application
in our setup we have da system for customers where speakers can dial in
with phonenumber+1 and the listeners dial in with phonenumber
the speakers conference is started with 323963...@conf+flags{waste}
the listeners conference is
I haven't really used waste much myself, but my understanding is that
waste and mute would conflict, since waste says send audio always
and mute says send audio never. I didn't understand why you're using
waste on the listeners... you should be able to get by with waste just
on the speaker (again,
thank you for your response
as a listener waste influences what you hear and mute say's you cannot speak
this is what our customer wanted because the speaker is the only one who is
heard in this conference or meeting room - this rooms are for lectures
we tried to disable waste for the listeners
waste + mute would result in sending audio that was all zeros or generated
silence.
On Tue, Sep 1, 2009 at 4:32 PM, Bradley Brashier bjbrash...@gmail.comwrote:
I haven't really used waste much myself, but my understanding is that
waste and mute would conflict, since waste says send audio
that means something in your path does not support CNG/VAD.
it's perfectly ok to use waste and mute together.
there is no chance that you would not enter the conf muted the way you
describe unless you are
using an older revision of FS that had a bug in the parsing of the
conference flags.
In the command to lists a conference it talks about member status, shown
as hear|speak|floor
What does the floor mean as a status - I can't find any reference to it
anywhere.
--
Alan Chandler
http://www.chandlerfamily.org.uk
___
FreeSWITCH-users
On Tue, Aug 18, 2009 at 4:30 PM, Alan Chandler
a...@chandlerfamily.org.ukwrote:
In the command to lists a conference it talks about member status, shown
as hear|speak|floor
What does the floor mean as a status - I can't find any reference to it
anywhere.
I just spoke with Mike J about
I didn't see any SIP session timers in the wiki. Since I'm already using the
event socket for control, my current plan is to use sched_api to play a file
with a short (20ms?) clip of silence, capture the play_file event and use it
to reschule another one for a couple of seconds later.
I'll let
That sounds horrible. There are settings both in sip/rtp and in
conference to do this already.
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#session-timeout
All right, I'm confused.
The RTP timeout parameters have no documentation, and from the names, I'd
have guessed that they hang up after a specified amount of time, not send
some other signal.
The session-timeout timer talks about calls expiring, and sending another
SIP invite, which I don't
Hi all.
The solution to this one should be short.
My conference hangs up when there's 2+ users and silence for 5 sec or so.
I'm trying to find a parameter that changes that (I'd rather it be, say, 60
seconds).
I didn't see a parameter like this specific to conferences, so I looked
abroad a bit.
Check out the 'waste' member flag. I think if at least one member has that
set then RTP will get sent out even during silence. Let us know if that
helps...
-MC
On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier bjbrash...@gmail.comwrote:
Hi all.
The solution to this one should be short.
My
I'm sure that would work, but I'm worried about it sucking up bandwidth,
especially since you'd need it on every caller (since otherwise the one
person who had it could hang up and you'd be back to square 1).
Any other ideas, or should I hunt through the code to try to override the
behavior
On Thu, Aug 13, 2009 at 12:47 PM, Bradley Brashier bjbrash...@gmail.comwrote:
I'm sure that would work, but I'm worried about it sucking up bandwidth,
especially since you'd need it on every caller (since otherwise the one
person who had it could hang up and you'd be back to square 1).
Any
My guess is that its the other end killing the call due to rtp
timeouts, not us killing the call. If you can confirm this the best
method would be to get them not to do rtp timeouts.
On Aug 13, 2009, at 1:47 PM, Bradley Brashier wrote:
I'm sure that would work, but I'm worried about it
I'm currently running current trunk synched up Tues morning, but it was
happening in all of the versions I'd been using previous -- I first
downloaded around the end of May.
I'll look into getting you a PCap. I expected that this was a known thing
with a parameter somewhere, so I haven't looked
I took a closer look at the SIP messages on the console. From it, I
understand that it's not Freeswitch timing out, but rather FS is getting the
BYE msg from somewhere else. I've tested phones and tested calling without
going through the FS conference, though, and everything works fine. Then I
saw
OK, I finally got a moment to do a packet capture and take a look at the
streams. It became very clear very quickly that what happens is that during
silence the gateway still sends RTP packets to Freeswitch, but Freeswitch
doesn't send any back to the gateway. After 10s of this, the gateway says
Hello!
Is any ability to ask to unmute in conference?
--
С уважением, Кривушин Михаил
Ведущий специалист отдела телекоммуникаций,
ООО РН-Информ филиал в г.Томске,
г. Томск сот. +7 913 865 78 66
icq: 218 744 127
xmpp: krivushi...@jabber.ru
mail: krivushi...@rn-inform.tomsk.ru
Внутренняя сеть:
Hello!
Is any ability to ask to unmute in conference?
--
С уважением, Кривушин Михаил
Ведущий специалист отдела телекоммуникаций,
ООО РН-Информ филиал в г.Томске,
г. Томск сот. +7 913 865 78 66
icq: 218 744 127
xmpp: krivushi...@jabber.ru
mail: krivushi...@rn-inform.tomsk.ru
Внутренняя сеть:
2009/7/9 Кривушин Михаил krivushi...@rn-inform.tomsk.ru
Hello!
Is any ability to ask to unmute in conference?
Not sure if I understand the question. Are you talking about the caller
pressing zero to mute/unmute?
-MC
___
Freeswitch-users mailing
mute/unmute is a toggle.
Mike
On Jul 9, 2009, at 11:20 AM, Michael Collins wrote:
2009/7/9 Кривушин Михаил krivushi...@rn-
inform.tomsk.ru
Hello!
Is any ability to ask to unmute in conference?
Not sure if I understand the question. Are you talking about the
caller pressing zero to
My guys want to work with operator - I wrote an WEB-face for conferencing.
And he wants to mute all participants, and give voice by order.
May be say caller_id on press any button to all conference.
I think to make execute_application + say caller_id. I will try to introspect
channel vars in
conference xyz mute all
On Thu, Jul 9, 2009 at 10:47 AM, Кривушин Михаил
krivushi...@rn-inform.tomsk.ru wrote:
My guys want to work with operator - I wrote an WEB-face for conferencing.
And he wants to mute all participants, and give voice by order.
May be say caller_id on press any button
I added support so when multiple endconf users are in the same conference
the total number of people with the flag must reach
0 before it kills the conf.
Can I take a small break now please ;)
That's why I'm afraid to add new stuff sometimes.
On Sat, May 30, 2009 at 9:16 PM, jcro...@gmail.com
No breaks! keep improving the conference app :)
--Stephen
On Mon, Jun 1, 2009 at 6:08 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
I added support so when multiple endconf users are in the same conference
the total number of people with the flag must reach
0 before it kills the
Ok, my _javascript_, dialplan, etc has been Wiki'd...
The new page I created [Examples _javascript_ Conference IVR] is
here:
http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR
I linked it from the [_javascript_ Examples] page here:
I would set the endconf flag in addition to the moderator flag so when
you leave it kicks everyone. :)
/b
On May 30, 2009, at 1:11 AM, jcro...@gmail.com wrote:
Regarding #3:
Here is what I'm used to... When the (last) moderator drops, the
users hear something like the moderator has left
Did someone add that feature? Are you messing with my head? =)
That would almost work perfectly... One scenario to consider -
sometimes, in a large group of folks that meet regularly, there are
multiple callers that come in as a moderator because they're not sure
who might not be able to
+flags{endconf|moderator}
Its there...
/b
On May 30, 2009, at 11:48 AM, jcro...@gmail.com wrote:
Did someone add that feature? Are you messing with my head? =)
That would almost work perfectly... One scenario to consider -
sometimes, in a large group of folks that meet regularly,
If my freeswitch box was accessible at the moment, I would try this
out... Unfortunately, it's not, so could you elaborate a little bit on
this scenario?
Imagine I had 4 people in a conference, two of them entered with
{endconf|moderator}, the other two did not.
What happens when ONE of the
I think I can answer my own question after looking at the code... It
seems that when THAT ONE user leaves, a flag is set that notifies the
conference thread to teardown the conference. I guess I will have to
roll my own on this one I guess, especially since I don't want to kill
the
I could not get this working on current trunk. Can you post your
configuration on conference module and the dialplan example?
Thanks,
jmesquita
On Thu, May 28, 2009 at 12:56 PM, Michael Collins m...@freeswitch.orgwrote:
On Wed, May 27, 2009 at 8:00 PM, j3flight jcro...@gmail.com wrote:
I'd be very interested to see your script and dialplan , for me it's a
very important issue as my conference server facing to outside and I
need to have moderators and regular users entering different pins.
Anton
jcro...@gmail.com wrote:
Unfortunately, the instance of FreeSwitch where I've
Maybe someone can do a wiki page with scripts and a howto?
/b
On May 29, 2009, at 3:31 PM, Anton Karpov wrote:
I'd be very interested to see your script and dialplan , for me it's a
very important issue as my conference server facing to outside and I
need to have moderators and regular users
I'll post the script, dialplan and how-to on the wiki as soon as I
can...
Will follow up on this thread with a link once it's complete.
Thanks for the interest.
Brian West wrote:
Maybe someone can do a wiki page with scripts and a howto?
/b
On May 29, 2009, at 3:31 PM,
On Wed, May 27, 2009 at 8:00 PM, j3flight jcro...@gmail.com wrote:
Wiki Tax paid...
That was my first contribution to the freeswitch wiki!
MC, you're welcome to have a look over it and see if i made things clear
enough. Feel free to edit.
Nicely done! Thanks for taking the time to create a
Thanks j3flight, I have used that method, but the profiles seem to be for
conferences not users. So I give a conference a profile, and everybody in
the conference shares the profile settings. If I'm wrong, let me know,
otherwise I'll stop hijacking this thread.
--Stephen
On Wed, May 27, 2009 at
Nope, it seems you are absolutley correct. I had setup my conferences this
way, but hadn't experimented yet. I tried putting two users into the same
conference using different profiles, but they both had the same caller
controls. Bummer.
I believe we can get around this though, assuming I can
It would be an improvement to move the caller controls to the member so it
works
the way you expect but it will have to wait for the right time and
motivation level.
From there it would also be possible to make up some new caller controls
that were moderator inspired like (mute all besides
Quoting Mr. Anthony Minessale:
thThe easiest way would be the new feature I added to 13442
in the conference profile add
param name=conference-flags value=wait-mod/
to your profile
and in your dialplan
action application=set data=conference_member_flags=*moderator*/
action
Wiki Tax paid...
That was my first contribution to the freeswitch wiki!
MC, you're welcome to have a look over it and see if i made things clear
enough. Feel free to edit.
On Tue, May 26, 2009 at 4:56 PM, Anthony Minessale
And the wiki tax if you feel comfortable adding this to the wiki. If
Jason,
There are many ways to accomplish this using FreeSWITCH. All of
which will require you to do a little bit of coding in js, lua or some
other language.
1. Park all callers into a fifo.. (see mod_fifo)
2. When leader auths in your script then you uuid_transfer them all
into
Ok, that sounds doable... I have no problem banging around with some
code. Thanks for the advice...
I'm new to FIFOs and FreeSwitch in general, so please check my logic
here...
- When a user enters their conference number, I check to see if that
conference exists (because the conference
First off, I apologize if this has been sent multiple times, the mailing
list won't cooperate with me... Hopefully that is resolved now.
I'm attempting to replicate the behavior of an Asterisk conferencing
system and I need a feature that, I'm surprised to say, doesn't seem to
be supported
I'm attempting to replicate the behavior of an Asterisk system with
FreeSwitch and I need a feature that, I'm surprised to say, doesn't seem to
be supported (easily).
Ok, so I've setup my dialplan so that when a specific extension is hit, it
calls out to some javascript which acts like an IVR to
the easiest way would be the new feature I added to 13442
in the conference profile add
param name=conference-flags value=wait-mod/
to your profile
and in your dialplan
action application=set data=conference_member_flags=moderator/
action application=conference data=1...@wideband/
or
action
On Tue, May 26, 2009 at 4:56 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
the easiest way would be the new feature I added to 13442
in the conference profile add
param name=conference-flags value=wait-mod/
to your profile
and in your dialplan
action application=set
Michael,
Pardon me for hopping on this thread, but can you explain more about this
new feature? I've been wanting something like this to apply different
behaviors for different conference members. Can this be used to provide a
'moderator' with different behaviors bound to DTMF keys than regular
Anthony - unbelievable! Thank you so much for implementing that! I kept
going through the possibilities of using a FIFO, putting the javascript in a
polling loop, or having everyone enter the conference muted and manually
playing MOH. This feature absolutely makes my code a snap... Yes, I
I don't *believe* there is anything of this nature built in at this
time. I would imagine it can be implemented. If you yourself don't
know C, but would like this functionality, three ways to get something
like this added would be:
A) Put up a bounty on the bounty page of the Wiki and hope
No, this doesn't exist yet in FreeSWITCH but I do like the idea.
/b
On May 21, 2009, at 3:17 PM, Yossi Neiman wrote:
Now that I've said that I don't believe it currently has these
features,
I'm waiting for Brian West to come in and correct me like he usually
does. :-)
Brian West
Hello, all. Being new to FS, I was curious if there are any logs/cdrs which
could be generated to gather statistics about a conference call? I'm mainly
looking for call duration and user count. So far, my CDR's only have
individual user CDR's, but nothing for a conference bridge.
Thanks!
That was it.
I installed the hd sounds and it works now.
Thanks.
Brian West schrieb:
Chances are he just doesn't have the 16k sound files installed.
/b
On Apr 7, 2009, at 6:35 PM, Michael Collins wrote:
2009-04-08 00:43:24 [ERR] mod_sndfile.c:194 sndfile_file_open() Error
I want to use a low bandwidth codec. But whenever I try to use speex I
get an error in the conference. We have FS trunk 1288. Switching back to
PCMx it works again.
Is there any problem with speex and DTMF or with transcoding?
Best regards
Peter
2009-04-08 00:43:23 [DEBUG] sofia_glue.c:2624
2009-04-08 00:43:24 [ERR] mod_sndfile.c:194 sndfile_file_open() Error
Opening File
[/usr/local/freeswitch/sounds/en/us/callie/conference/conf-pin.wav]
[System error : No such file or directory.]
2009-04-08 00:43:24 [WARNING] mod_conference.c:4799
conference_function() Cannot ask the user
Chances are he just doesn't have the 16k sound files installed.
/b
On Apr 7, 2009, at 6:35 PM, Michael Collins wrote:
2009-04-08 00:43:24 [ERR] mod_sndfile.c:194 sndfile_file_open() Error
Opening File
[/usr/local/freeswitch/sounds/en/us/callie/conference/conf-pin.wav]
[System error : No such
If any can share some ideas, I'm looking at making conference calls simple
for the end-users of my FS system.
Here are some issues I'm kicking around:
1. To create/join conferences - do you make a pre-defined list of
extensions, each of which would join the caller to a particular conference
Steven Ward pisze:
If any can share some ideas, I'm looking at making conference
calls simple for the end-users of my FS system.
Here are some issues I'm kicking around:
1. To create/join conferences - do you make a pre-defined list of
extensions, each of which would join the caller to
Szymon, I want to provide a service wherein a user can reserve a
teleconference room for a partiuclar time and control who can join the
conference call (only those invited). I want to support several of
such conference calls at any given time.
I want callers who were not invited to a conference
Steven Ward pisze:
Szymon, I want to provide a service wherein a user can reserve a
teleconference room for a partiuclar time and control who can join the
conference call (only those invited). I want to support several of
such conference calls at any given time.
I want callers who were
Hey, if you guys get this all figured out, tested, and working then
please be sure to put it on the wiki. You could create a whole new
page and then link to/from the mod_conference page.
-MC
On Wed, Mar 25, 2009 at 7:02 AM, Szymon Olko so...@gcdf.pl wrote:
Steven Ward pisze:
Szymon, I want to
I need some time to work out my setup and explore some different options,
but I'll be happy to get something together for the wiki on this as soon as
I'm able.
Thanks.
On Wed, Mar 25, 2009 at 4:53 PM, Michael Collins m...@freeswitch.org wrote:
Hey, if you guys get this all figured out, tested,
Hmmm no MOH wont work... since I am planning on pulling more than just 2
members into the conference and I still need ringback for the later
members as well.
Is there a direct way for me to use conference number play audiofile
to play teletone directly? or should I just records some ringing if I
There is a file format called tone_stream that I was trying to explain
yesterday.
tone_stream://teletone spec
or
tone_stream://path=/path/to/text_file.ttml
you can use this to play tones anywhere a filename is supposed to go.
I guess loopback really is your only option if you must generate
Hmm ok ... Ill try that In my head though the api call to originate
shouldent block? but I assume since it does my head is wrong.
Thanks you for the explanation. I think you can put this one to bed now
:-P
On Tue, Feb 03, 2009 at 07:54:29AM -0600, Anthony Minessale wrote:
There is a file
hmmm ok indeed.
small mods to js files to just play a lng tone_stream full of ringy
noises and then stop them in the on answer and I have what I wanted.
Thank you very very much for all your help.
On Tue, Feb 03, 2009 at 04:16:21PM +0200, Sias Mey wrote:
Hmm ok ... Ill try that In my head
Loopback will not work in that case either. If the far end plays
ringback inband you should hear that if you use the conference dial
api call.
/b
On Feb 2, 2009, at 4:24 AM, Sias Mey wrote:
Aaah ok.
Thanks for clearing that up.
So using loopback is still the only real workable
Actually loopback does work.
however as I said it generates a pair of extra channels.
Hmmm I was trying to generate and extra call to a JS script that
generated a teletone ring in an on_ring_execute for the second call
however it seems to stop execution of the call itself. Event though I
use api
Hi,
Im trying to build a web based conference control system.
Got most of it sorted with some help from the list but I seem to have
run into some strangeness.
I use a conference dial call to pull extra users into the conference.
I couldent find a way of setting channel variables or executing
What wasn't working about this? The {} can be used everywhere without
a problem... Maybe you can provide more details on this.
/b
On Jan 30, 2009, at 4:39 AM, Sias Mey wrote:
I couldent find a way of setting channel variables or executing
javascript directly on the conference dial
Hi Brian,
Hmmm Ill do some more testing on it later. But I got a destination out
of order when I tried. Right now Im busy implementing the string
checking. Which seems like it will work out ok, but is clearly not
ideal.
Thanks for the replay
On Fri, Jan 30, 2009 at 04:45:54AM -0600, Brian West
you should be able to use {} in the dial command
you also should be able to do
originate
{...}sofia/profile/u...@domain.comconference:conf_name@profile_name
inline
to the api interface
On Fri, Jan 30, 2009 at 7:33 AM, Sias Mey s...@cpdata.co.za wrote:
Hi Brian,
Hmmm Ill do some more
On a similar note is it possible to use api commands from the dialplan.
I would like a execute_on_answer to run a script in the same fasion, but
I cant seem to get it to execute as a api command.
___
Freeswitch-users mailing list
Hmmm no it didnt... but at least now I know what to fix when it doesent
work whe I update again.
Thank you very much for your help.
Sias
On Fri, Jan 23, 2009 at 08:13:16AM -0600, Anthony Minessale wrote:
That was the change i checked into trunk to allow app::arg as well as
appspacearg
Hmm ok... updated to the latest SVN and tried your suggsestion however
all I can see happening in the console is
2009-01-23 11:35:44 [ERR] hangup.js:2 mod_spidermonkey()
ReferenceError: request is not defined
(obviously I renamed foo.js to hangup here)
thanks again for the help.
On Thu, Jan
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