Daqiang:
How do you make your IP phone not dial right after you press #? Usually
the IP phone will dial the number already once you pushed #
On Sun, Oct 11, 2009 at 10:45 AM, daqiang wang wangdq@gmail.com wrote:
it's work . Thank you very much .
2009/10/11 Michael Collins
Dear All,
I'm trying to use FS with xml_curl as routing server for otgoing calls to my
provides. Here is simplified setup:
[REGISTRARs] [APP SERVERs] [ROUTING SERVER/FreeSwitch]
[SBC/no Transcoder] [Terminating GWs/ITSPs]
Because of simplicity I do not use hunting
this is up to your phone # means address complete and you phone sends
the number you dialed into an INVITE message.
if you want to support FAC with # you should modify the phone's dialplan and
make it expect more digits... for certain prefixes.
T.
On Sun, Oct 11, 2009 at 12:10 PM, Henry
Hi all!
As it stays in wiki:
...
HEARTBEAT
Status information for freeswitch trigerred by freeswitch's heartbeat every
20 seconds.
...
Is there any way to customize timeout of HEARTBEAT events?
Thanks in advance,
Artem
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Did anyone find a solution to this problem? I too would like to install the
esl module for PHP.
Herman
Harondel J. Sibble wrote:
Okay, just finished installing 1.0.4 from tarball on Ubuntu 9.0.4 server,
then went to install FreePBX v3, I've gotten all the prerequisities in the
wizard
Hi Eric -
The way that we do it is to keep each gateway in its own Sofia profile.
Issuing
api sofia profile profile start reloadxml
does one call to the web server for that profile's XML, which can be
pretty compact if it just contains one gateway.
--Dave
Hello,
We are looking at
On Oct 7, 2009, at 10:48 AM, srinivasula reddy wrote:
Hi
can any please tell me where registered calls are stored, so when
incoming call came to mod_sofia.c how it will check it is registered
or not?\\
Calls are not registered and calls have nothing to do with
registration. Users
Although probably not the best solution, I figured out a way to make it
compile and install:
I removed all of the -Werror instances in
PATH_TO_FREESWITCH_SOURCE/libs/esl/Makefile
If I was a hardcore c/c++ programmer, I'd figure out the real problem.
Herman aka frek818
On Sun, Oct 11, 2009 at
I'd like to add this for the next weekly conference.
I have added a few events to the event list, as you can see here:
http://wiki.freeswitch.org/wiki/Event_list
But I need more help from the community to complete that and add content to
the events, etc. So if you can add that for the next
Maybe you want enable_heartbeat or sched_heartbeat from mod_dptools?
You can pass your parameters in second to these two.
Example:
action application=enable_heartbeat data=1/
action application=sched_heartbeat data=1/
Where 1 in this case is the number of heartbeats per seconds.
You can use
On Oct 9, 2009, at 2:41 AM, Gabriel Gunderson wrote:
On Fri, Oct 9, 2009 at 12:08 AM, Michael Collins
m...@freeswitch.org wrote:
Thanks for reporting back. Please let all the Asterisk users know
that they
are welcome to join us in #freeswitch on irc.freenode.net and that
they will
On Oct 9, 2009, at 10:40 AM, Maciej Aniserowicz wrote:
Hello,
The issue is resolved. I feel stupid, because Michael Jerris was
right the first time. Setting external_rtp_ip and external_sip_ip to
$${local_ip_v4} made it work.
But the strange thing is: it SOMETIMES worked before without any
There is this endless push and pull on this topic, those who want them
assume it should be default, those who don't assume that should be
default. This probably needs a configuration option defaulting to
pass them (those who don't want to pass them are usually a bit more
educated and
We don't have session messages directly exposed, except for things
like display, respond, and deflect. What specifically are you trying
to send ?
Mike
On Oct 9, 2009, at 3:34 PM, Matthew Fong wrote:
I'm used to using the onInput callbacks inside lua and javascript to
listen for dtmf
I have seen this on the wiki too, for example:
Q: Does it require hardware, kernel modules, ztdumshit, etc?
Nope! :D You must be thinking of [http://sofaswitch.org/eg/aac.jpg|Something
Else]
I know that's just a joke and we might make one or two jokes, but we don't
really hate Asterisk and
can you confirm from an rtp packet trace that they are all really
sending 20ms?
Mike
On Oct 10, 2009, at 6:04 AM, Maciej Aniserowicz wrote:
Hi,
Here are the messages with a:ptime parameter. All the calls are
started by
commands sent through socket.
I'm not sure if this is all
On Oct 11, 2009, at 5:44 PM, Diego Viola wrote:
Maybe you want enable_heartbeat or sched_heartbeat from mod_dptools?
You can pass your parameters in second to these two.
Example:
action application=enable_heartbeat data=1/
action application=sched_heartbeat data=1/
Where 1 in this case
I am still working on the new build system for esl, stay tuned for
more info soon, it should be in 1.0.5.
Mike
On Oct 11, 2009, at 5:36 PM, Herman Griffin wrote:
Although probably not the best solution, I figured out a way to make
it compile and install:
I removed all of the -Werror
Mike,
Thanks for getting back to me. I agree.
I'm willing to throw down on a bounty for this. Any idea how much
work we're talking about here?
On Sun, Oct 11, 2009 at 5:59 PM, Michael Jerris m...@jerris.com wrote:
There is this endless push and pull on this topic, those who want them
No we do not issue a subscribe OUTBOUND. We work with Polycom, Snom
and a few other that outbound subscribe is something we should do if
we want to know you took the phone off hook.
/b
On Oct 9, 2009, at 10:23 AM, Jerry Richards wrote:
I gather from the mailing archive that BLAs are
btw the polycom it will do the sub like you expect but the rest will
only know when the phone is actually on a call or receiving a call...
we won't know if the handset is taken off hook...I would like to
rework the functionality similar to how sofia_sla.c handles it.
Again if you want to
Well since we aren't a proxy you shouldn't default to passing them
right?
/b
On Oct 11, 2009, at 6:12 PM, Kristian Kielhofner wrote:
Mike,
Thanks for getting back to me. I agree.
I'm willing to throw down on a bounty for this. Any idea how much
work we're talking about here?
I have tried to police the wiki when things like this appear.. its one
thing to crack a joke in fun from time to time... but to put stuff
like that on the wiki isn't acceptable.
/b
On Oct 11, 2009, at 5:06 PM, Diego Viola wrote:
I know that's just a joke and we might make one or two
FreeSWITCH can play back stereo files it'll just mux them down to mono
before playing... can you elaborate on the error you're getting?
/b
On Oct 10, 2009, at 12:40 AM, Seven Du wrote:
Yes, it's discussed before.
http://wiki.freeswitch.org/wiki/Channel_Variables#RECORD_STEREO
set that var
got
2009-10-12 01:41:30.349961 [WARNING] switch_core_file.c:133 File has 2
channels, muxing to mono will occur.
2009-10-12 01:41:30.349961 [ERR] switch_core_codec.c:431 Stereo is currently
unsupported. please downsample audio source to mono.
freeswi...@internal
freeswi...@internal version
Seven Du dujinf...@gmail.com wrote:
originate {ignore_early_media=true,RECORD_STEREO=true}sofia/gateway/xx/xx
bridge(sofia/gateway/yy/yy)
Shouldn't that be record_stereo=false for mono recording?
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Brian,
You are correct, they *probably* shouldn't be passed by default.
On Sun, Oct 11, 2009 at 7:44 PM, Brian West br...@freeswitch.org wrote:
Well since we aren't a proxy you shouldn't default to passing them
right?
/b
--
Kristian Kielhofner
http://www.astlinux.org
I set to true because brian said it can play stereo files but no lucky for
me.
2009/10/12 Jason White ja...@jasonjgw.net
Seven Du dujinf...@gmail.com wrote:
originate {ignore_early_media=true,RECORD_STEREO=true}sofia/gateway/xx/xx
bridge(sofia/gateway/yy/yy)
Shouldn't that be
Where are you playing the files?
/b
On Oct 11, 2009, at 9:27 PM, Seven Du wrote:
I set to true because brian said it can play stereo files but no
lucky for me.
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not sure I said this but open a jira... its a bug thats recent...
usually it will mux them to mono and the codec engine is trying to
open L16 with two channels so something has changed to cause this
regression.
Expect a fix sometime tomorrow.
/b
On Oct 11, 2009, at 9:27 PM, Seven Du
Is it possible to play a stereo file ?!!! whats the conclusion.?!
Brian West-3 wrote:
Please open a jira please this did work but a recent change in
switch_core_codec caused this to appear I usually test this regularly
but haven't run thru a full run of tests lately.
/b
On Oct 11,
Hi Mike,
Thanks for your valuable reply,
when i install freeswitch1.0.2 in my machine(Windows xp operation system) i
dont have any databasae installed in my system, then from sqllite will come
into picture, and how can i see the registered users data from sqllite.
Thanks
Srinivas
On Mon, Oct
It was possible but we have a regression in the code that isn't
letting that happen right now... hence the reason i said Open a jira
so we could fix it.
IS THAT not clear?
/b
On Oct 11, 2009, at 10:46 PM, Nagalenoj wrote:
Whats the conclusion.?!
Michael, Diego,
thanks for the rapid answers!
As far as I understand, enable_heartbeat app is launching
SESSION_HEARTBEAT events that will stop when the call will be cleared. Also
I heard that enable_heartbeat works only for calls with proxied media.
What I want is to monitor FreeSwitch status:
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