Conferences are started dynamically when the first participant enters,
and end when there are no participants. There is no need to start a
conference before any participants enter.
Mike
On Jun 26, 2008, at 3:24 PM, e schmidbauer wrote:
is there anyway to start a conference automaticaly
You can likely do this with a combination of broadcast (see
uuid_broadcast api command) and mod_tone_stream which lets you use
teletone just like you are playing a file.
Mike
On Jun 22, 2008, at 10:49 PM, Klaus Teller wrote:
Hi Folks,
I was was wondering if it's possible to play a
If you want to actually not send to them you need to set the effective
caller id name/number.
Mike
On Jun 20, 2008, at 1:10 PM, Brian West wrote:
It will always contain the number. Its the far ends responsibility to
honor the privacy flags. The same happens on a PRI as far as I have
I believe it does, we should re-try the send in this case. Can you
file a bug for this on jira (with a patch would make it even better :D )
On Jun 19, 2008, at 12:43 PM, Jonas Gauffin wrote:
Hello
I got some problems with the event socket, I do not receive all
events.
I've confirmed
There were problems with it in all of the rc's but it should be fine
in 1.0.0
Mike
On Jun 18, 2008, at 6:44 PM, UV [EMAIL PROTECTED] wrote:
It’s quite embarrassing but I read about all the great things mod_xm
l_rpc can do, see all the threads about it, even manage to load the
module
Any bay area freeswith users interested in catching up for a drink
sometime this week, please drop me an e-mail off list.
Mike
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
Contact me offlist and can we arrange for me to get into your box
remotely.
Mike
On Jun 13, 2008, at 8:01 PM, Areski K wrote:
Anyone have been working with an MSSQL ODBC connection through
Javascript FS !?
I am facing this constant error :
2008-06-14 01:38:22 [CRIT] switch_odbc.c:240
It does not at this time. We have looked at adding support for rtcp
when the standards for doing rtcp on the same ports as rtp are
solidified.
Mike
On Jun 2, 2008, at 8:29 AM, Peder @ NetworkOblivion wrote:
Does FS support RTCP? I am interested in getting per call quality
stats
from
.
Let me know how can I help troubleshoot it.
If there's a way I can disable the Allow-Events headers, I'll be
able to
confirm my hypothesis.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Michael
Jerris
Sent: Monday, June 02, 2008 3:15 AM
Is this freeswitch on both sides of the call here? Can i get into the
bod live to trouble shoot this?
Mike
On Jun 1, 2008, at 6:35 AM, UV wrote:
I've checked, double checked and triple checked. Both configurations
are
identical.
Take a look at the pastebin capture:
at 8:09 AM, Michael Jerris [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Currently the directory interface is only used for that
dialplan, I
would like to enhance that in the future. The directory
dialploan
uses a filter of exten=destination number, and then has name
I reviewed your bug on jira. Your mera is sending a 500 error.
Please look into this problem with whoever provides support for your
mera.
Mike
On Jun 2, 2008, at 12:59 AM, Pieter Eduard wrote:
Hey all,
thanks for the reply, appreciate it so much,, my fs version is
FreeSwitch Version
Currently the directory interface is only used for that dialplan, I
would like to enhance that in the future. The directory dialploan
uses a filter of exten=destination number, and then has name/value
pairs, I will see if I can find the schema we used back when we
developed it, short of
We fixed this in make sofia-reconf late yesterday. Please update and
try again
On May 22, 2008, at 12:38 PM, RR wrote:
On Sat, May 17, 2008 at 12:29 PM, Brian West [EMAIL PROTECTED]
wrote:
This is a requirement if you're following trunk:
svn update
make sofia-reconf current vm-sync
Hi
.
Updated to revision 8528.
after this make sofia-reconf current vm-sync still gives the same
error. Same is the case on make current
Thx
On Thu, May 22, 2008 at 1:27 PM, Michael Jerris [EMAIL PROTECTED]
wrote:
We fixed this in make sofia-reconf late yesterday. Please update
and try again
As stated before, this is the WRONG way to set this up. Please setup
sangoma as TDM-api and use span_wanpipe.
Mike
On May 21, 2008, at 10:13 AM, Helmut Kuper wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
hm, I still can't find a reproducable way for setting up FS with
I have done a very basic msi that visual studio builds, but it does
not include bootstrapper for the runtime or the sound files. If
someone has good experience with doing a more full installer, I would
love to have them take a look.
Mike
On May 21, 2008, at 1:53 PM, jeff sacksteder wrote:
Typically sip handsets have a reject button. If there is not a button
for this on your handset, then there is no way to do it from that
handset.
Mike
On May 19, 2008, at 9:34 AM, Czaderna wrote:
Hello
I've got question about rejecting a calls. I make a call between two
wireless phones
We don't have any real sizing numbers on this, but I would guess that
having your location for recorded messages on ram disk would make a
big difference. A bit of sipp magic with pcap playback will probably
give you some real world numbers and we would love to hear how much
you can scale.
DOH!! I knew I forgot something. Will roll those this morning. You
can grab the rc2 ones, I think there was only a couple small changes.
On Apr 17, 2008, at 6:55 AM, Peder @ NetworkOblivion [EMAIL PROTECTED]
wrote:
OK, I got through the make and install with rc3, but now I am trying
I think with the volume of calls you are handling, this is one place
where openser will serve you better than freeswitch. You said you
already have openser in this role, why would you not want to use it?
Mike.
On Apr 16, 2008, at 1:09 PM, kokoska rokoska [EMAIL PROTECTED]
wrote:
That being said, there is an open bug that the tarballs only work with
automake 1.9. That will be fixed in the next rc.
Mike
On Apr 16, 2008, at 10:54 PM, Anthony Minessale [EMAIL PROTECTED]
wrote:
rc2 and up is pre bootstrapped.
On Wed, Apr 16, 2008 at 10:46 PM, Peder @
I think you will need to try to tweak the build for unicode support to
fix this.
Mike
On Apr 15, 2008, at 11:48 AM, Jonas Gauffin wrote:
assert is made on while (*s *s != '' (*s != '%' || t != '%')
!isspace((int) (*s)))
*s contains ösvan
Full stack trace:
We have not added any MSC type functionality at this time.
Mike
On Apr 11, 2008, at 2:42 PM, Anya wrote:
Hi all,
can someone tell me if this project supports IuUP interface? I am
mainly
interested in support for Iu-CS.
I am looking for a small MSC that supports A and IuUP interfaces
that that is different between our environments. I will fix up the
inet_pton issues tomorrow (later today) and let you know.
Mike
On Apr 10, 2008, at 1:57 AM, Tamas wrote:
Hi,
As I wrote in previous mail, it was svn trunk r8070.
Regards,
Tamas
Michael Jerris írta:
Is this with svn trunk (what
I believe by spec its 2* the number of physical channels, but don't
quote me on that. Specifically thats for everything going on on those
spans including suspended calls (which we don't currently support
anyway)
Mike
On Apr 9, 2008, at 4:08 PM, Andy Spitzer wrote:
Woof!
On Wed, 09 Apr
error LNK1181: cannot open input file
'..\..\..\..\w32\library\debug\freeswitchcore.lib'
Regards,
Tamas
Michael Jerris írta:
What os/compiler versions?
Mike
On Apr 9, 2008, at 2:16 PM, Tamas Cseke wrote:
Hello,
I have 2 problems:
1, libvoipcodecs
C2143: syntax error : missing
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