the 1.0.9 sounds were rolled tonight and they contain these fixes.
Mike
On May 5, 2009, at 8:38 AM, Peter P GMX wrote:
I looked at my install directory and in the source files
(freeswitch-sounds). No file of this name there.
Thanks for the link. Now it works.
Best regards
Peter
Brian
Hello,
I have to static IP:s on my server. FS has been bound to one of them.
Yesterday evening I got these log messages:
2009-05-05 19:17:37 [INFO] mod_sofia.c:2938 general_event_handler() IP
change detected [85.89.XX.XX9]-[85.89.XX.XX8] []-[]
2009-05-05 19:17:37 [NOTICE] sofia_glue.c:3303
Hi Seven,
I am exactly looking for this functionality.
Please let me know when you are finished with new queue manager app. I'll
try it in my call center.
Regarding Patch: is it already part of SVN trunk? If not then could you help
me how to install it, I have no programming background.
Many
add param name=auto-restart value=false/ to conf/
autoload_configs/sofia.conf.xml
Did you happen to bind the IP while FS was running?
/b
On May 6, 2009, at 1:36 AM, Jonas Gauffin wrote:
Hello,
I have to static IP:s on my server. FS has been bound to one of them.
Yesterday evening I got
The patch haven't been merged into trunk. It should be as easy as
execute the following command in the FS source code root dir:
patch /tmp/the_patch_file_name.diff
I will post an example on the wiki when I finished, hope be soon.
On May 6, 2009, at 6:45 PM, Saeed Ahmed wrote:
Hi Seven,
I
No, have never changed the IPs since the server was installed. And have not
changed it in FS either.
Ok. Will add the parameter. Thanks.
On Wed, May 6, 2009 at 2:01 PM, Brian West br...@freeswitch.org wrote:
add param name=auto-restart value=false/
to conf/autoload_configs/sofia.conf.xml
Did
The IP guessing code changed its guess then...
/b
On May 6, 2009, at 7:57 AM, Jonas Gauffin wrote:
No, have never changed the IPs since the server was installed. And
have not changed it in FS either.
Ok. Will add the parameter. Thanks.
Brian West
br...@freeswitch.org
-- Meet us at
chenexyee wrote:
1. user A is in conversation with user B, and at this time, a incoming
call from user C comes to A, in this case, I want freeswitch to play
busytone to C, how to configure?
you could use the limit app (mod_limit) to limit A's number of calls to
1, then play the busy sound with
I worked on the patch and added it to trunk rev 13240
On Wed, May 6, 2009 at 7:53 AM, dujinfang dujinf...@gmail.com wrote:
The patch haven't been merged into trunk. It should be as easy as execute
the following command in the FS source code root dir:
patch /tmp/the_patch_file_name.diff
I
FYI,
Please update your installations as soon as possible. More information on
this update is available here http://www.freeswitch.org/node/181.
Thanks for all of your feedback - please keep it coming and join us on IRC
if you have any questions about the newest version.
-Michael S Collins
On May 6, 2009, at 9:47 PM, Raymond Chandler wrote:
chenexyee wrote:
1. user A is in conversation with user B, and at this time, a
incoming
call from user C comes to A, in this case, I want freeswitch to play
busytone to C, how to configure?
you could use the limit app (mod_limit) to
The exact same way you use it for outbound... just use limit before
you call the user in your dial plan. An inbound call to a user is
nothing more than an outbound call from FreeSWITCH to the user.
/b
On May 6, 2009, at 9:11 AM, dujinfang wrote:
I'm also finding a way to limit only one
Thanks Guys
_
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Wednesday, May 06, 2009 3:57 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Inboud Call Queue
Thanks, so quick. Actually I had submitted another version of patch
which added a channel var fifo_caller_exit_to_orbit which make the
caller possible to exit to the orbit_exten other than hangup the
caller when the caller enter the fifo_caller_exit_key.
I use this to guide the caller to
Hi All,
Has anyone tried Zanzibar and Cairo. They have implemented MRCP 2.0
and integrated Sphinx 4. Since MRCP 2.0 supports SIP for
communication, it can easily be integrated with FS!
Although it is implemented in Java, VoiceXML is also supported!
--
Regards,
Moiz Chinoy.
You're a bit mistaken on the MRCP 2.0 supporting SIP.. it uses SIP for
signaling and RTP for media transport. ... that however doesn't mean
you can just call it via SIP and it work. They put a little bit more
goop on top of that!
/b
On May 6, 2009, at 9:58 AM, Moiz Chinoy wrote:
Hi
Yes you are right, it uses SIP for signaling and RTP for media
transport. But does it make any difference because in its
documentation they have stated that they support Asterisk and any IP
PBX that supports SIP and RTP.
I haven't tried it with FS yet but it worked with Xlite using SIP. For
some
FYI,
Here's a nice story http://www.freeswitch.org/node/182 for you all to
check out. Please check it out and pass it on.
-Michael
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
On Wed, May 6, 2009 at 8:23 PM, Michael Collins m...@freeswitch.org wrote:
FYI,
I made a comment on Dave's blog extolling the virtues of FS and I mentioned
Skype support. I didn't specifically mention mod_skypiax but I didn't
specifically mention any mods.
blushI was suggesting to put
Are the most currently ratified HD Voice codecs G.722 and G.722.1? I haven't
heard very much about HD Voice at all until you just brought it up.
From: Michael Collins m...@freeswitch.org
To: freeswitch-users@lists.freeswitch.org
Well its not so much which codecs but that the codecs can do 16k...
Currently FreesWITCH supports d...@16k, sp...@16k, G722 and g72...@16k
But as a bonus:
We can also do sp...@32k, g722...@32k, c...@32k and 48k
/b
On May 6, 2009, at 1:38 PM, Paul wrote:
Are the most currently ratified HD
Using the default installation, I've noticed that when I (or someone
else) calls in on my SIP trunk and keys in an extension, not all of
the numbers are recognized unless they hold the key down for at least
1/2 second to a second.
Is there a way to improve DTMF recognition so people can
Well it depends.. first off are you doing inband dtmf or RFC2833?
Secondly what SVN rev are you running?
/b
On May 6, 2009, at 1:44 PM, Jay Austad wrote:
Using the default installation, I've noticed that when I (or someone
else) calls in on my SIP trunk and keys in an extension, not all of
I'm running 1.0.4pre3. Haven't gotten a chance to upgrade to pre7 yet.
2833 is the default right? I haven't changed anything. I'm using
voicepulse for my SIP trunks. Is there an option I can add to that
definition to force RFC2833?
--
jay austad | 612.423.1433 | aus...@signal15.com
On Wed, May 6, 2009 at 11:35 AM, Giovanni Maruzzelli gmar...@celliax.orgwrote:
On Wed, May 6, 2009 at 8:23 PM, Michael Collins m...@freeswitch.org
wrote:
FYI,
I made a comment on Dave's blog extolling the virtues of FS and I
mentioned
Skype support. I didn't specifically mention
Hi Jay,
Have to say my DTMF works flawlessly on thousands of calls. (SVN trunk
from a couple of days ago. We handle around 100,000 calls/day via FS)
That said, I've found it depends on your SIP trunk provider.That
doesn't mean to say there isn't a problem; it's just that I haven't
Date: Wed, 6 May 2009 09:47:42 -0400
From: intralan...@freeswitch.org
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Busy tone and text message configuration
chenexyee wrote:
1. user A is in conversation with user B, and at this time, a incoming
call
See this:
http://wiki.freeswitch.org/wiki/Simple_call_center_using_mod_fifo
On May 6, 2009, at 10:15 PM, Saeed Ahmed wrote:
Thanks Guys
From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org
] On Behalf Of Anthony Minessale
Sent:
Hello all,
Help a n00b out.I have been trying to get an instance of FreeSwitch
running up in the Amazon EC2 cloud.
I have successfully gotten the package built following the wiki and archives
of this list.
I can get x-lite to register with the switch and it will set up calls
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