Hello,
today I uploaded a little patch for openzap into trunk (r667). It marks
now inbound channels as inUse which is conform with outbound channel
handling. This should solve some problems finding a free channel in
ozmod_isdn.c for inbound and outbound calls.
regards
Helmut
Hello,
just to keep you informed about this problem. As mentioned I added a
hack to free allocated channels depending on last event time. I enhanced
oz dump as well to display last event time and InUse-Flag. What I
found is this:
1.
InUse channel flag wasn't set for inbound calls. I fixed that
Andrew,
I think you're right, packets are indeed sent to 172.31.0.13 while
mod_erlang_event is listening at 127.0.0.1 ! Why didn't I see that ! ;-)
Will test it now and let you know how it goes..
regards,
Leon
On Feb 24, 2009, at 1:22 AM, Andrew Thompson wrote:
Leon,
I think I found the
Well, this works, I feel a bit stupid now :-] Now it's time to play
with it..
Thanks a lot !
kind regards,
Leon
On Feb 24, 2009, at 1:22 AM, Andrew Thompson wrote:
Leon,
I think I found the problem. I shouldn't have been defaulting to
binding
to 127.0.0.1, instead the default should
Hi,
the clue for sending fax is to use the originate command in the CLI:
originate sofia/example/1...@10.10.10.10 txfax(/path_to_fax_file)
this command will send the fax file via profile example to fax machine
100 reachable via 10.10.10.10
Hope this could help others :p
regards,
rod.
I was reading through the Skypiax documentation and saw the comment
that it's not possible to run multiple skype clients on the same linux
machine, all using the same skype user account.
It's possible to run multiple skype clients with the same skype user
account, as long as the skype
Please report this bug to jira.freeswitch.org.
On Feb 24, 2009, at 2:27 AM, Rene Pankratz wrote:
No, unfortunately the problem still persists. Portaudio still
automatically accepts/takes the next call.
René
On Thu, Feb 19, 2009 at 5:04 AM, Rene Pankratz
r.pankr...@fh-wolfenbuettel.de
What direction is the original call?
Are you sure you do not have the auto_answer enabled?
On Tue, Feb 24, 2009 at 1:27 AM, Rene Pankratz
r.pankr...@fh-wolfenbuettel.de wrote:
No, unfortunately the problem still persists. Portaudio still
automatically accepts/takes the next call.
René
On
On Tue, Feb 24, 2009 at 10:49:24AM +0100, Leon de Rooij wrote:
Well, this works, I feel a bit stupid now :-] Now it's time to play
with it..
Nah, bad choice of defaults on my part. Defaulting to 0.0.0.0 is much
more consistant and compatible. For some reason I was trying to emulate
the event
Hi,
The file directory.conf.xml had been mentioned in the documentation many times
but there is not such file in the conf folder. Do you mean default.xml in
directory folder?
Thanks!
--- On Tue, 2/24/09, freeswitch-users-requ...@lists.freeswitch.org
Hello.
After upgrade to version 1.0.3 we have a problem with the codec iLBC
(I think that this is due to the transition to a new ilbc libs 1 week
ago).
Very poor quality for calls to the codec iLBC mode=20 (crack in the
dynamic). iLBC mode=30 works well.
Tested with phones and Zoiper SJPhone.
The problem comes up that the default is 30... the chances are that
your phone doesn't set the mode= line so we default to 30 when this
takes place. Not setting the mode= line in the FMTP usually means
30ms... which is the default. So to force this always to 30 you can
allow i...@30i,
Hi,
I have a small javascript application that accepts a call, retrieves some dtmf
digits and then records the call to an icecast server. This works great.
The problem I'm having is that when the call is being recorded freeswitch is no
longer sending rtp packets back to the originating
On Tue, Feb 24, 2009 at 9:24 AM, Ali Al-Rubaie kerrada2...@yahoo.com wrote:
Hi,
The file directory.conf.xml had been mentioned in the documentation many
times but there is not such file in the conf folder. Do you mean default.xml
in directory folder?
Thanks!
Can you tell me where you see
is it during a bridged call?
On Tue, Feb 24, 2009 at 11:49 AM, Dan freeswitch-us...@digitaldan.comwrote:
Hi,
I have a small javascript application that accepts a call, retrieves some
dtmf digits and then records the call to an icecast server. This works
great.
The problem I'm having is
Hello
Maybe this question has been raised before, but if not: There's so
much traffic in this mailing list that I was wondering if adding a
web-based forum on the site was in the works?
Cheers,
___
Freeswitch-users mailing list
Maybe this question has been raised before, but if not: There's so
much traffic in this mailing list that I was wondering if adding a
web-based forum on the site was in the works?
We are upgrading the freeswitch.org site soon to drupal 6.9. We are
considering turning on the forum feature
The web version of this list is available at:
http://www.nabble.com/Freeswitch-users-f32209.html
Mike
On Feb 24, 2009, at 2:08 PM, Fred wrote:
Hello
Maybe this question has been raised before, but if not: There's so
much traffic in this mailing list that I was wondering if adding a
I'm getting this error message trying out the pizza demo in FS 1.0.3:
?Can't open dictionary C:\Program Files\FreeSWITCH\grammar\default.dic
I didn't have this before where there was no default.dic file.
Is there some place a path has to be set now?
Thanks.
http://www.bkw.org/pizza_gram.tar.gz
/b
On Feb 24, 2009, at 1:36 PM, mszla...@aol.com wrote:
I'm getting this error message trying out the pizza demo in FS 1.0.3:
Can't open dictionary C:\Program Files\FreeSWITCH\grammar
\default.dic
I didn't have this before where there was no
Hi Brian,
It sounds like I'd be better off with 1.0.3 than SVN and will waiting for the
fix?
But thanks for the files and info.
Mark.
-Original Message-
From: Brian West br...@freeswitch.org
To: freeswitch-users@lists.freeswitch.org
Sent: Tue, 24 Feb 2009 11:45 am
Subject:
On Tue, Feb 24, 2009 at 4:19 PM, Michael Collins m...@freeswitch.org wrote:
Maybe this question has been raised before, but if not: There's so
much traffic in this mailing list that I was wondering if adding a
web-based forum on the site was in the works?
We are upgrading the
hi all,
i come from asterisk an i am new to freeswitch. after my with days with
freeswitch i am very excited!
but trying to migrate our deployment i have three challenges. one of them is:
i need to call freeswitch from a webapp (e.g. python) and pass number1 and
number2. i then need
On Tue, Feb 24, 2009 at 1:09 PM, Alexander de Greiff
alexan...@degreiff.com wrote:
hi all,
i come from asterisk an i am new to freeswitch. after my with days with
freeswitch i am very excited!
Welcome to FreeSWITCH!
but trying to migrate our deployment i have three challenges. one of them
Hey Brian,
Where abouts do you keep the Window MSI 1.0.3 build that isn't in SVN trunk.
Installing from the wiki installation page gets me a build with the same error.
Thanks. Mark.
-Original Message-
From: Brian West br...@freeswitch.org
To:
On Mon, Feb 23, 2009 at 5:26 PM, Carlos Talbot carlos.tal...@gmail.com wrote:
Were you planning to check in the sample skype.conf.xml into the default
FreeSWITCH conf folder? If so, just be aware the default config causes
freeswitch to hang right after a load mod_skypiax (if you do not have
On Tue, Feb 24, 2009 at 10:33 AM, Eric Chamberlain e...@rf.com wrote:
I was reading through the Skypiax documentation and saw the comment
that it's not possible to run multiple skype clients on the same linux
machine, all using the same skype user account.
It's possible to run multiple skype
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