Just seen Anthony presentation, very cool ;)
Everyone, watch it!
http://files.freeswitch.org/cluecon_2009/presentations/Day%2001%20Presentation%2002.Anthony%20Minessale.1500kbps.mp4
=D
On Wed, Aug 12, 2009 at 5:07 PM, Michael Collins m...@freeswitch.org wrote:
On Wed, Aug 12, 2009 at 2:25
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
http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference
by using application=conference and then the data string as
Hey Michael,
Just wondering something, I have found that you added
conference_set_auto_outcall on the dptools wiki, but I could not find that
function in the mod_dptools.c, shouldn't that be part of the mod_conference
wiki article? =D.
Best regards,
Diego
On Wed, Aug 12, 2009 at 1:50 PM,
Hello all.
I want to use Grandstream Early Dial future.
How i can enable support 484 response?
I tried simply use
action application=hangup data=484/
and
action application=respond data=484/
on uncompleted extensions,
but there is not work
Thanks.
Yuriy .
Hi,
Googeling about this shows that FS aims to support this (in fact it supports
all 3: UDP/TCP/TLS).
Yet I could not find the way to configure FS in order to support that.
In fact, it does not work in my current install.
I have TLS configured and work, but could not make TCP works
thanks in
It probably belongs there. It's a wiki, feel free to fix it. What
does this have to do with this thread?
On Aug 13, 2009, at 4:03 AM, Diego Viola diego.vi...@gmail.com wrote:
Hey Michael,
Just wondering something, I have found that you added
conference_set_auto_outcall on the dptools
Hi,
I am new to FreeSWITCH and need an advice.
All calls to PSTN from our server will go through single gateway,
which is a soft switch supporting SIP protocol. FreeSWITCH will need
to register with soft switch, but soft switch permits only single
active call (in either direction) per
It just works... to force TCP you append ;transport=tcp
In reality you should be using SRV records.
/b
On Aug 13, 2009, at 3:40 AM, Tzury Bar Yochay wrote:
Hi,
Googeling about this shows that FS aims to support this (in fact it
supports all 3: UDP/TCP/TLS).
Yet I could not find the way
I don't think we ever got this working correctly. Can you do a trace
of it working vs not working?
/b
On Aug 13, 2009, at 3:14 AM, Yuriy Ivzhenko wrote:
Hello all.
I want to use Grandstream Early Dial future.
How i can enable support 484 response?
I tried simply use
action
It just works... to force TCP you append ;transport=tcp
Hi Brian
In fact this is exactly what I did and I could not get it work.
I am using a console based application supplied by pjsip.org and when
trying to register i get some error messages saying 'invalid transport
SIP/2.0/tcp' and 'REGISTER
I need to see the sip packet. TCP should be uppercase I'm pretty sure.
/b
On Aug 13, 2009, at 9:04 AM, Tzury Bar Yochay wrote:
nta: Via check: invalid transport SIP/2.0/tcp from
80.74.97.189:42634
nta: REGISTER (30669) has invalid Via
___
Hello!
First of all, I would like to express my thanks to all the developers of
Freeswitch.
I am testing Freeswitch on a Debian machine with physical network
interface with four virtual IP addresses. One of these IP addresses,
aliased as eth0:3, has been created specifically for Freeswitch.
If you read the latest vars.xml I have clarified this:
!--
THIS IS ONLY USED FOR DINGALING
bind_server_ip
Can be an ip address, a dns name, or auto.
This determines an ip address available on this host to bind.
If you are separating RTP and SIP
On Aug 13, 2009, at 9:37 AM, Tzury Bar Yochay wrote:
I need to see the sip packet.
dumped below
TCP should be uppercase I'm pretty sure.
you mean the via should be Via: SIP/2.0/TCP right?
Yep
If so, then that would a bug in the client then.
Some things might accept it but sofia is
Thanks Ken.
I'll look at mod_limit
The XLite softphone doesn't seem to have a switch for controlling it.
-str
On Aug 12, 2009, at 9:22 PM, Ken Rice wrote:
Check out mod_limit... Other wise you'll have to look specifically
at the UA
you are trying to use, some like polycom and sipura offer
Thank you Michael,
I will tinker around with it and definitely follow-up with the results.
- T
___
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On Wed, Aug 12, 2009 at 7:19 PM, Brian West br...@freeswitch.org wrote:
Well you really can't ignore it... it happens with our ISDN stack
too. Thats what the VETO handles.
/b
You lost me. What do you mean we can't ignore it? the way I see it, sure we
can and we should.
Currently that
On Thursday 13 August 2009 16:47:18 Brian West wrote:
I don't think we ever got this working correctly. Can you do a trace
of it working vs not working?
I can't do working trace, only not working
http://pastebin.freeswitch.org/9980
with dialplan action
action application=hangup
So it sounds like set can work. But you'd still have to parse it. And even
then it's not recommended.
I have another couple of possible methods for you:
1) modification of mod_conference.
2) event socket.
If you modify mod_conference, you can probably do what you want, but it
obviously requires
On Thu, Aug 13, 2009 at 2:35 AM, Timur Irmatov irma...@gmail.com wrote:
Hi,
I am new to FreeSWITCH and need an advice.
All calls to PSTN from our server will go through single gateway,
which is a soft switch supporting SIP protocol. FreeSWITCH will need
to register with soft switch, but
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
I couldn't imagine managing a conference without a GUI. I need to see who is
making noise so I can boot/mute em ;)
If I were you I would dive into ESL and make a simple web app to frontend
the conferences. There will surely be something in contrib to get you
started.
On Thu, Aug 13, 2009 at 8:48
Err, I asked if that was wrong to fix it.
On Thu, Aug 13, 2009 at 2:15 PM, Diego Viola diego.vi...@gmail.com wrote:
I was talking with Michael about fixing stuff in the wiki, so I just asked
to fix that also.
On Thu, Aug 13, 2009 at 9:10 AM, Michael Jerris m...@jerris.com wrote:
It
Thanks for responding, Michael!
On Thu, Aug 13, 2009 at 9:06 PM, Michael Collinsm...@freeswitch.org wrote:
Are you going to have incoming calls as well? If so, how does the
soft-switch handle two concurrent calls to the same number?
Yes, we'll have incoming calls as well. I did not performed
Hey Michael,
I am a little late to the party I know - but just want to say thanks
for your latest efforts.
I updated my dev environment with the latest managed mod and swapped
my app to the latest plugin architecture last night and all is working
well.
Love the dynamic loading of my dll into
Hi Michal,
Just checking in to see if you've been able to take a stab at this.
Thanks,
Vladimir
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michal Bielicki
Sent: Tuesday, August 11, 2009 5:33
I've been trying to bridge FreeSWITCH with a Toshiba CIX using qsig
over a PRI. I have a Sangoma A102 card installed in a Dell PowerEdge
with CentOS 5.3. The issue I am having is no packets are being
transmitted back to FreeSWITCH. ifconfig w1g1 shows every frame
received as an overrun. I've tried
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
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
It probably just VETO it so it avoid sending
SWITCH_MESSAGE_INDICATE_PROGRESS again since the call is already
making progress from the core's point of view?
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 13-Aug-09, at
Yes, agreed, but there is no point in sending a WARNING since is a normal
condition, therefore should not even try to change the state of the channel.
On Thu, Aug 13, 2009 at 7:57 PM, Mathieu Rene mrene_li...@avgs.ca wrote:
It probably just VETO it so it avoid sending
64bit OpenSolaris w/ gcc-4.3.2
After a bootstrap and configure I get the following error when running make:
---snip---
Compiling src/switch_caller.c ...
cc1: warnings being treated as errors
src/switch_caller.c: In function 'switch_caller_profile_event_set_data':
src/switch_caller.c:299:
The reason it works when you wait 3 seconds is that mod_loopback bails
out of the equation as soon as it detects a bridge.
It'll call switch_ivr_uuid_bridge() and you'll end up with 2 sofia
channels.
Is there a hook that is fired when when that switch_ivr_uuid_bridge()
successfully executes?
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