What method are you trying to use?
If you're using the event socket, it's in several of the parameters.
For instance, with an add-member event, you could look at
Caller-Username, Caller-Caller-ID-Name, or Caller-Caller-ID-Number
depending on your specific needs. There's also a uuid available if
Has anyone given this any thought? Do I need to provide more information?
It's still not making any sense to me, and I'm planning on just
removing all of the default dialplans, but I'd like to make sure this
won't recur in the future.
BB
On Fri, Sep 25, 2009 at 9:33 AM, Bradley Brashier bjbrash
Hi guys,
I've got a strange situation that I'm at a loss to explain. With all
callers, I go through a dialplan where I check to see if they should
be a moderator, then transfer them to another which puts them into a
conference accordingly. This worked great on one server, but when I
copied the
, Bradley Brashier bjbrash...@gmail.com
wrote:
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
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
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,
Speaking as someone who went through this recently myself, my
suggestion is to learn by doing. Choose some simple-sounding task you
want to accomplish in FS, and try to carry it out. Use the wiki as a
reference and the mailing list if you can't figure out what you need
from that.
I don't suggest
Well, you'd have another nickel from over here, then.
If I can get this working before I'm tasked with something else I'll write
up something more on the wiki about Freeswitch and SIPp, but I'm not sure
I'll get that chance.
BB
On Tue, Aug 25, 2009 at 11:05 AM, Anthony Minessale
exactly what you said you
didn't want to do.
Mike
On Aug 13, 2009, at 6:24 PM, Bradley Brashier wrote:
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
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-hold-timeout-sec
http://wiki.freeswitch.org/wiki/VAD_and_CNG
Mike
On Aug 14, 2009, at 2:41 PM, Bradley Brashier wrote:
I didn't see any SIP session timers in the wiki. Since I'm already using
the event socket for control, my
.
Good luck with whatever you end up doing.
BB
On Thu, Aug 13, 2009 at 12:11 AM, Alan Chandler
a...@chandlerfamily.org.ukwrote:
Bradley Brashier wrote:
I wrote:
This is a significant new fact for me. What you seem to be doing is
calling the commands referenced in the conference api here
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.
manually?
BB
On Thu, Aug 13, 2009 at 9:50 AM, Michael Collins m...@freeswitch.org wrote:
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
into it too terribly far
myself, yet. I'm gonna try looking at the console outputs and logs myself,
first.
BB
On Thu, Aug 13, 2009 at 12:44 PM, Michael Collins m...@freeswitch.orgwrote:
On Thu, Aug 13, 2009 at 12:47 PM, Bradley Brashier
bjbrash...@gmail.comwrote:
I'm sure that would work
on it today. I'll do that sometime when the system is less busy.
BB
On Thu, Aug 13, 2009 at 1:45 PM, Bradley Brashier bjbrash...@gmail.comwrote:
I had just thought of the exact same thing. I'm trying to test that now.
Thanks for your input.
BB
On Thu, Aug 13, 2009 at 1:20 PM, Michael
I'm back to looking at Freeswitch to figure out how to send just a
single packet every second or so during silence. If anyone knows of a way to
do this, let me know, otherwise I'll get back to you if and when I find one.
BB
On Thu, Aug 13, 2009 at 2:48 PM, Bradley Brashier bjbrash...@gmail.comwrote
You've thought through some of the difficult points, which is good. You're
right that the moderator can't have different controls (unless you're
controlling the conference yourself from outside, using, say, the event
socket).
Before I go further, I want to make sure I understand what you're
Whoops. All of my parens () should be curly braces {}. Wasn't paying
attention.
BB
On Wed, Aug 12, 2009 at 2:40 PM, Bradley Brashier bjbrash...@gmail.comwrote:
You've thought through some of the difficult points, which is good. You're
right that the moderator can't have different controls
:33 PM, Alan Chandler
a...@chandlerfamily.org.ukwrote:
Bradley Brashier wrote:
...
Before I go further, I want to make sure I understand what you're
proposing. What you're essentially saying is that when the command to
kick someone is pressed the person should be transferred out
for transcripting.
timer-name
Specifies the name of this profile's timer. To separate it from other
timers?
BB
On Fri, Jun 19, 2009 at 8:59 AM, Bradley Brashier
bjbrash...@gmail.comwrote:
OK, I figured out the TTS stuff. It's a matter of choosing an engine (I
chose flite), uncommenting a few things
, Bradley Brashier
bjbrash...@gmail.comwrote:
So it turns out that it wasn't a bug at all -- it is a feature that was
not implemented. So I've got some work to do to get that running. Since I
said I would, though, here's my analysis of the conference parameters you
were asking about:
mute
what I have.
On Wed, Jun 17, 2009 at 9:07 PM, j3fli...@gmail.com wrote:
FYI: I fixed the Wiki documentation for the lock/unlock feature.
Bradley Brashier wrote:
So I found one interesting thing so far: the lock caller control
actually does function as a toggle, and, in fact, unlock does
at it again in a month or so when the project is closer to done.
But if anyone has any ideas on why certain phones would behave worse than
others (a Polycom SoundPoint IP 320 SIP phone is the worst) I'm all ears.
BB
On Tue, Jun 16, 2009 at 2:58 PM, Bradley Brashier bjbrash...@gmail.comwrote:
Will do, just
confusing for us new users.
Can we add some documentation in there to that effect, perhaps?
BB
On Thu, Jun 18, 2009 at 7:26 AM, Bradley Brashier bjbrash...@gmail.comwrote:
What I did last night was to go ahead and modify mod_conference.c to
include a new count conference control. I've got it getting
I was indeed looking at announce-count, but from the code, it looks like
that was designed to announce to the caller how many people were on the
conference only when they were joining and the number was over a threshold
specified in the profile. Not exactly what I was looking for, but it did
help
I've been using multiple digits successfully right from the start, about 2
or 3 weeks ago. They do the separation of *1 and *10 the same way as several
other systems -- by time. If you dial *, then 1, then wait past a timeout,
then 0, you'll get *1, and *10 if you did it faster. I've tested by
Actually, that's another good reason to do those wiki and/or code
comments changes... most likely, the reason you thought it couldn't be done
is that you tried it and it didn't work... but you tried it on the default
profile before you realized that it was hard coded. I know that's what I did
and
Well, since what I really need at this time is only about 5 commands of
similar complexity to a toggle on something already extant, I've decided to
just modify source. I can't imagine that people will be terribly interested
in my modifications, but I know I'm interested in being able to stay
dialplan from the caller controls in
conferences a while back. Depending on your goal, that might be an easy way
to get your problem resolved. You can keep state using the hash api and
hash on the conference name or some other useful thingie.
On Wed, Jun 17, 2009 at 1:18 PM, Bradley Brashier
I'm creating a conferencing product for use in a system with theoretically
several hundred concurrent calls. I'm using FreeSwitch to create this
product, but am not only new to FreeSwitch, but also the entire telecom
industry as well as Open Source projects in general (I'm a recovering BIOS
guy).
voice traffic is your network infrastructure. If
you can isolate FS and some phones on a separate, controlled network then
possibly you can start narrowing it down to other factors.
-MC
On Tue, Jun 16, 2009 at 10:51 AM, Bradley Brashier bjbrash...@gmail.com
wrote:
I'm creating
:
freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Bradley
Brashier
*Sent:* Tuesday, June 16, 2009 1:52 PM
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* [Freeswitch-users] Voice lag in conference
I'm creating a conferencing product for use in a system
How much power do I have with DTMF conference controls? The wiki doesn't
have much information on this. For example, one of the things I'd like to do
is take the currently existing lock and unlock actions and merge them
into a lock toggle action. Preferably in XML configuration files. Is this
even
.
*From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Bradley
Brashier
*Sent:* Tuesday, June 16, 2009 5:02 PM
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] Voice lag in conference
...@freeswitch.org wrote:
Bradley Brashier wrote:
How much power do I have with DTMF conference controls? The wiki
doesn't have much information on this. For example, one of the things
I'd like to do is take the currently existing lock and unlock
actions and merge them into a lock toggle action
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