THE BLOODY MADNESS!!! I can only stop if people start saying 'NO'. :)
/b
On Nov 3, 2009, at 1:21 PM, Anthony Minessale wrote:
Don't forget the one where there was a typo in the one for G722 so
now we are all required to emulate that typo by running a 16khz
codec with 8khz timestamps and
At some point the paint will be rubbed off the magic lamp.
/b
On Nov 3, 2009, at 1:11 PM, Kristian Kielhofner wrote:
It appears that Tony has already added an option (amazing) BUT you
should really be setup for central provisioning with an installed base
that large... You'll eventually have
Yah this one is LLLAME :P
We have some dyslexic engineers.
/b
On Nov 3, 2009, at 1:12 PM, Arsen Chaloyan wrote:
Another issue is G726 bit packing. Again some implementations used
wrong bit packing and RFC3551 tried to partially resolve this
conflict introducing new payload format
Yes you can login and edit the wiki yourself.
Thanks,
/b
On Nov 3, 2009, at 9:16 PM, Dmitry Gromov wrote:
Hi!
Was just reading wiki here: http://wiki.freeswitch.org/wiki/Home_PBX_Example
It lists sample sofia.conf.xml which has this parameter:
!--param name=inbound-no-media value=true/--
I
you know I have heard this before... It seems to ONLY be ATT
/b
On Nov 2, 2009, at 9:54 AM, Ujjval Karihaloo wrote:
Yes, I think I did. However here is what furthur testing revelas. If
I dial in from ATT cell phone, I do not see any DTMF using Don's
IVR.xml.conf to call my conf app. But
Is starpound involved in the FS Community?
/b
On Nov 2, 2009, at 12:51 PM, Artem Shiyanov wrote:
Here is rather big and, let's say, complete example of mod_java usage:
https://starpound.svn.sourceforge.net/svnroot/starpound/trunk/src/fs2agi
The goal of this project is to be a proxy between
So you're doing your billing INLINE with the session using the OSBC
Query app? Not the most optimal way!
/b
On Oct 31, 2009, at 10:59 AM, Rupa Schomaker wrote:
I believe this means you are hung up in CDR reporting. You can get a
core of the running system and look at the stack traces to
You should never do billing inline with the session thread is all I'm
saying.
/b
On Oct 31, 2009, at 11:32 AM, Dome Charoenyost wrote:
I use odbc_query for retrive balance and get LCR from my billing DB.
and use nibble_bill
Dome C.
___
I think once you get the backtrace like rupa said we can see that
maybe odbc_query is really hanging or something similar.
/b
On Oct 31, 2009, at 12:05 PM, Dome Charoenyost wrote:
2009/10/31 Brian West br...@freeswitch.org:
You should never do billing inline with the session thread is all
feature for us.
Regards
Brian Stafford
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http
You have to be doing it wrong then.
Can you show us your dialplan you should have two extensions one for
the lot range and one to attended transfer someone into the lot.
/b
On Oct 30, 2009, at 10:47 AM, Brian Stafford wrote:
Hi all
I did a 'make update' to 15289 and I found the auto
Brian West wrote:
You have to be doing it wrong then.
Can you show us your dialplan you should have two extensions one for
the lot range and one to attended transfer someone into the lot.
/b
The relevant excerpt from the dialplan is
extension name=valet_unpark
condition field
First you forgot to mention what SVN rev you're on...
/b
On Oct 30, 2009, at 5:07 PM, Humberto Quintana wrote:
Hi everybody,
I'd like some help with this situation that is 'haunting' me :-)
My scenario is as follows:
inbound-bypass-media is set in the profile because we dont want FS
No its not ALL of the sudden... Please check the SRTP settings on the
identity. Thats why you're getting that... the phone defaults to
sending crypto in AVP which is invalid... we reject it.
This has been this way for some time I suspect you have updated your
snom recently or aren't on
SEE THIS TAG? If you have SRTP on you MUST put this to Optional if
you care for it to work... read the debug a little better it'll tell
you to reference an RFC on this also.
/b
On Oct 29, 2009, at 7:40 PM, Lars Zeb wrote:
RTP/SAVP: off
perl -i -pe 's/ss7_boost/sangoma_boost/g' freeswitch.spec
or svn up.
/b
On Oct 29, 2009, at 9:22 PM, Joseph L. Casale wrote:
I fetched a copy of trunk, tar and gzipped it up and placed it in my
SOURCES dir, I edited the bzip extension in the spec file and tried to
build rpms but got this
Please update to 1.0.5pre4 if you plan on using the CELT codec on the
conference tomorrow. I'm rolling the pre4 tarball right now.
Thanks,
Brian
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org
Type conference at the cli and the help will be displayed... same
can be called over XML RPC.
/b
On Oct 27, 2009, at 11:51 PM, Lon Baker wrote:
Just like you can call conference list for conferences, is there a
way to retrieve the profile state of a conference member using the
cli or
http://wiki.freeswitch.org/wiki/Channel_Variables#playback_terminators
Are you using sync or async socket?
/b
On Oct 28, 2009, at 3:29 AM, velusamy velu wrote:
Dear All,
I have played the list of voice files in playback like the
following by using ESL perl module,
Yes it is the expected behavior... if you wish to set them on an
inbound call TO the user you'll use the set_user api to do so. Its
kinda like sudo in FreeSWITCH it'll load up all the variables for that
user.
/b
On Oct 28, 2009, at 3:33 AM, mayamatakeshi wrote:
I can see the channel
http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core
Please don't cross post if possible.
/b
On Oct 28, 2009, at 4:07 AM, srinivasula reddy wrote:
Hi,
can i use sqlserver instead of sqllite. and can two freeswitch
servers can share same database(sqllite or sqlserver). any help
would
What kind of router are you behind?
/b
On Oct 27, 2009, at 10:58 AM, Lars Zeb wrote:
Thanks for the reply, Brian.
Did something in FS change between v15183 and v15225 to make this
occur? I ask because this same configuration worked OK in the
earlier version.
Lars
have you updated to the latest SVN?
/b
On Oct 28, 2009, at 11:35 AM, mayamatakeshi wrote:
I'm seeing a strange issue with eyebeam behind NAT.
If I make a call from eyebeam to FS, I can see FS receives packets
from eyebeam's nat address (confirmed with tcpdump and
mod_event_socket DTMF)
Sleep 1000 ms... we usually bring up media too fast before the other
end is ready.
/b
On Oct 28, 2009, at 3:37 PM, Jerry Richards wrote:
I notice that when I call IVR from the PSTN, the Welcome to
Freeswitch...
introduction is clipped at the beginning, so it sounds like come to
sessions per second now/max
/b
On Oct 28, 2009, at 4:58 PM, Cliff Wells wrote:
current sessions, but I'm unsure of what the 0/30 means.
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
The issue is you're not behind nat-pmp or upnp so it can't figure out
what your public IP is... you'll have to fill that in manually or
enable those options on your router if possible.
/b
On Oct 26, 2009, at 10:03 PM, Lars Zeb wrote:
I haven’t changed anything since v15183, where it
Let me review that and see I can't off hand think of anything that
would cause that.
/b
On Oct 27, 2009, at 10:58 AM, Lars Zeb wrote:
Thanks for the reply, Brian.
Did something in FS change between v15183 and v15225 to make this
occur? I ask because this same configuration worked OK
From FreeSWITCH you can't force and endpoint to register... thats the
endpoints job... what you're doing is just flushing the reg out of
FreeSWITCH so now it can't find the endpoint... Try adding a reboot
on there maybe i'll reboot the device and cause it to register again...
/b
On Oct
you should have just needed to set the rtp-ip and the sip-ip on both
profiles to their values and it would have worked fine... their would
have been no need to set the ext-*-ip equiv.
/b
On Oct 27, 2009, at 8:14 PM, Lei Tang wrote:
Thanks Eliot, It works.
Ok then something is broken badly... and that makes NO sense. Because
it works on my box.
/b
On Oct 27, 2009, at 8:34 PM, Lei Tang wrote:
Hi Brian, It doesn't work if I only set rtp-ip and sip-ip. when I
set the ext-rtp-ip, it works fine.
2009/10/28 Brian West br...@freeswitch.org
you
No you can't remove them... And they are 100% valid so your SBC is in
the wrong.
/b
On Oct 26, 2009, at 12:05 AM, Ujjval Karihaloo wrote:
Hi,
I used the downlaoded TAR ball and my calls worked, however, when
upgrading to the SVN release...my SBC is rejecting the 200 OK (when
the FS
You'll have to do your own load testing. Nobody can really tell you
exactly how many you'll get.
/b
On Oct 26, 2009, at 10:39 AM, Ujjval Karihaloo wrote:
With the following spec for CPU and Memory can someone help me
guesstimating how many simultaneous calls and Calls/sec a FS server
Finding the exact rev that broke it would be helpful.
/b
On Oct 26, 2009, at 12:06 PM, Peter Olsson wrote:
Hmm... I remembered incorrectly about my setup :) The Avaya PBX
talks TLS to the Avaya SES Server, and then UDP to FS, not TCP -
sorry, my bad!
However, something that has changed
Bet your hardware just barfs on those like others have... I mean
really I HATE SIP. This is stupid.
/b
On Oct 26, 2009, at 12:39 PM, Peter Olsson wrote:
In the non-working one I don't have these, and instead I have these
headers;
X-FS-Display-Name: 9099
X-FS-Display-Number: 9099
Get a dedicated DSL line. They aren't that expensive... I have four
of them at my house!
/b
On Oct 26, 2009, at 12:37 PM, Lars Zeb wrote:
Is there a write-up anywhere that might help me with this problem,
or lacking that, can anyone offer advice?
At some point we'll have to NO NO NO fix your broken crap. :P The
reason we have sip so broken now is NOBODY CAN SAY NO!... Say it with
me... NO!
/b
On Oct 26, 2009, at 1:27 PM, Peter Olsson wrote:
I understand your frustration :) We deal with SIP integration with
about 10 different
I highly doubt it... You can wait for someone to post their results
but in the end you'll have to do your own load testing because not
everyone's numbers will jive with your use case. Which is the reason
the project never posts or endorses a set call count.
/b
On Oct 26, 2009, at 2:50 PM,
You have SIPit, which was the SIP Backoff till Pillsbury got their
panties in a wad.
/b
On Oct 26, 2009, at 3:03 PM, Michael Collins wrote:
I was wondering... does anyone make a SIP certification program
kinda like a pen-tester except to find all the ways your SIP setup
is broken? Just
yo quiero taco bell
/b
On Oct 26, 2009, at 7:04 PM, Michael Collins wrote:
Or at least a chalupa.
-MC
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
you're behind nat and you didn't set the ext-rtp-ip or ext-sip-ip
correctly?
/b
On Oct 26, 2009, at 9:29 PM, Lars Zeb wrote:
I have tried to update (make current) twice since 15183. All inbound
calls are picked up but the caller hears nothing but a couple of
clicks. The most recent
Well first off you're not defining a pine here...
confn...@profilename+flags{mute|deaf|waste|moderator}+[conference pin
number]
That might be why its not asking for a pin.
/b
On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
entry action=menu-exec-app digits=1 param=conference
Have you started moving the code into our SVN and using our
ticketing / issue tracker to help you manage issues?
/b
On Oct 22, 2009, at 9:13 AM, Georgiewskiy Yuriy wrote:
On 2009-10-22 16:04 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre
...:
TCOn Thu, Oct 22, 2009 at 3:59 PM,
AS per the email you and I exchanged we created the account and the
mod_h323 folder in endpoints
/b
On Oct 22, 2009, at 9:34 AM, Georgiewskiy Yuriy wrote:
hm, you not tell me what account created, and i don't try to do this.
___
Im going to guess its because mydomain.localhost doesn't resolve
outside the machine itself so the softphone never ends up knowing wtf
to do.
/b
On Oct 22, 2009, at 10:42 AM, freeswitch noob wrote:
I have a noobish question about setting up FS.
I have it installed and running.
I setup
I can't get what exactly you re talking about. Can you clarify ? Also
please include the packets of interest only not the full trace if its
not relevant to the bug.
/b
On Oct 22, 2009, at 10:44 AM, Helmut Kuper wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Mike,
here it is:
What are you using to make this call?
/b
On Oct 21, 2009, at 6:58 AM, ineya ineya wrote:
Codecs are fine. I spent much time experimenting with codecs and
completely missed, that freeswitch is modifiyng the SDP record.
When phone A is making a call the SDP contains candidate media
Your SVN Account will be done soon and the directory in endpoints is
already created for you to start importing your work.
Thanks,
/b
On Oct 21, 2009, at 7:24 AM, Georgiewskiy Yuriy wrote:
On 2009-10-21 13:39 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre
...:
TC2009/10/21
uuid_displace with mux option.
/b
On Oct 21, 2009, at 6:11 AM, Joey Carter wrote:
Hello
How can I add background music that will play during call?
Thanks
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
This will be fixed soon. Watch SVN.
/b
On Oct 21, 2009, at 11:45 AM, Keith Laaks wrote:
Hi,
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
none specifically exist... good ole trusty editor?
/b
PS: http://www.cudatel.com
On Oct 21, 2009, at 1:37 PM, Jerry Richards wrote:
Can anyone recommend a good 3rd party dialplan tool that will work
with
Freeswitch?
Best Regards,
Jerry
___
Well how are you trying to dial users?
/b
On Oct 20, 2009, at 9:16 AM, ineya ineya wrote:
No, it's the same thing with /64 on both ends.
I tried to built SVN version and modified just IP addresses in
sip_profiles, but still can't call from one phone to another. It goes
straght to voicemail.
The defaults will NOT work with ipv6 out of the box because the
sofia_contact on the directory only looks at the internal profile NOT
the internal-ipv6 profile... open up the directory default and change
the sofia_contact to prepend the internal-ipv6/u...@domain
/b
On Oct 20, 2009, at
Or just set the var to what you want it to say?
/b
On Oct 20, 2009, at 11:19 AM, Michael Collins wrote:
Under what conditions did you see unknown? I'm wondering if the
user can just pick a default other than unknown if he wants
something else to be displayed.
Thoughts?
-MC
Fix this or set it to silence.
/b
On Oct 20, 2009, at 3:57 PM, Kristian Kielhofner wrote:
EXECUTE sofia/s2s/+19412848...@65.196.170.129 playback
(local_stream://moh)
2009-10-20 20:54:19.112927 [ERR] switch_core_file.c:116 Invalid file
format [local_stream] for [moh]!
I think you just have it misconfigured because if ipv6 was broken I
would have William and Jason jumping on me.
Can you post your xml configs for the profile please?
/b
On Oct 19, 2009, at 7:16 AM, ineya ineya wrote:
Hi,
I installed the phonenix release, and changed IPv4, IPv6 addresses
Please review this:
http://wiki.freeswitch.org/wiki/Mod_event_socket#sendmsg
SendMsg uuid
call-command: execute
execute-app-name: one of the applications
execute-app-arg: application data
/b
On Oct 19, 2009, at 11:42 AM, Nikita Belov wrote:
And what event name to use for sendevent command?
On Mon, Oct 19, 2009 at 5:13 PM, Brian West br...@freeswitch.org
wrote:
I think you just have it misconfigured because if ipv6 was broken I
would have William and Jason jumping on me.
Can you post your xml configs for the profile please?
/b
OK what makes you think it failed? The fact you don't hear it?
/b
On Oct 19, 2009, at 7:41 PM, Seven Du wrote:
not sure about this, but did you try send dtmf to uuid 723f3dbb-
b87b-4cd4-98fc-
698eed7f2bdb other than cced4b9a-b6de-4be1-8c12-
3d18cc6e8454 ?
Bill has confirmed it works fine in latest trunk. So i'm not sure
what exactly you're doing wrong can you provide me some debug output
of what you're doing in your dialplan and the console debug?
/b
On Oct 19, 2009, at 3:51 PM, ineya ineya wrote:
Here is internal.xml
Nope its part of SIP it'll retry if a response isn't received in a
certain time.
/b
On Oct 18, 2009, at 10:29 AM, Juan Backson wrote:
Hi,
Is there anyway to configure freeswitch so that it won't retry the
SIP message when 200 OK is not received?
thanks,
jb
You can also try this
http://wiki.freeswitch.org/wiki/Channel_Variables#progress_timeout
/b
On Oct 18, 2009, at 10:29 AM, Juan Backson wrote:
Hi,
Is there anyway to configure freeswitch so that it won't retry the
SIP message when 200 OK is not received?
thanks,
jb
How are you using this from your javascript code?
/b
On Oct 17, 2009, at 7:13 PM, Adam Wilt wrote:
2009-10-17 23:56:21.848930 [ERR] switch_odbc.c:188 STATE: IM002 CODE
0 ERROR: [unixODBC][Driver Manager]Data source name not found, and
no default driver specified
2009-10-17
They all use the same core API. I suspect since our stun server is
just a cname to many public services out there you're just thinking
its dingaling related but its really not...
/b
On Oct 17, 2009, at 7:10 PM, Mark Campbell-Smith wrote:
I noticed a similar behaviour by issuing the stun
Its in /etc/ or where ever your default is.. is FS running as root?
/b
On Oct 17, 2009, at 10:16 PM, Adam Wilt wrote:
So, where is FreeSWITCH looking for the odbc.ini file?
I keep reading conflicting information, including /usr/local/
freeswitch/etc/, /usr/local/etc, /usr/etc,
If you setup your own stun server it wouldn't do that But the
hostlookup only solves half the problem .. getting the external IP vs
poking holes for RTP which is what stun will do.
/b
On Oct 15, 2009, at 10:35 PM, Mark Campbell-Smith wrote:
Thanks Brian. Is this something
You can signup for a wiki account and correct the documentation so
others don't repeat the same issue.
Thanks,
Brian
On Oct 16, 2009, at 12:16 PM, Jerry Richards wrote:
Okay, I think the contrib folder moved up one level. So the Wiki
installation documentation should probably be updated
no clue it depends on how you have it configured. If you have an
outbound proxy set it'll send all calls to the proxy. If not then
most likely it will go direct to the destination.
/b
On Oct 15, 2009, at 12:00 AM, Lars Zeb wrote:
Brian,
Does this mean that Bria should be sending
Sounds like bria is sending the invite via the proxy which is your
local FS box and NOT direct.
/b
On Oct 14, 2009, at 4:26 PM, Lars Zeb wrote:
I am trying to make a test call to the FreeSwitch conference
address, sip:8...@conference.freeswitch.org on a Bria softphone.
However, it fails
Why exactly are you doing proxy media?
/b
On Oct 14, 2009, at 4:56 PM, Klaus Hochlehnert wrote:
Does anyone have a solution for this?
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
Make current will NOT touch your configs and is designed to prevent
build skew.
/b
On Oct 14, 2009, at 5:14 PM, Klaus Hochlehnert wrote:
The conf and the scripts dir were untouched after that (which I
wanted because I didn’t want to reconfigure everything).
Update to latest trunk and try again... then if it persists then
collect all the sip traces and debug logs as per the wiki on reporting
bugs.
Thanks,
Brian
On Oct 12, 2009, at 5:01 PM, Klaus Hochlehnert wrote:
P.S.: I’m using FS trunk from last week and newest Snom Firmware
The sip_registration table contains the contacts for each registered
endpoint. Its not for the directory from a database per se... If you
wish to serve up your users and groups from a database check out the
XML Curl wiki page.
/b
On Oct 12, 2009, at 11:01 PM, srinivasula reddy wrote:
Does anyone see a problem with hosting mod_h323 in our SVN? I would
like to centralize everything we can to reuse our issue tracking
resources and not fragment the community if possible.
/b
On Oct 12, 2009, at 2:43 PM, Tihomir Culjaga wrote:
hi,
finally i compiled it right ... had a
Thats called mod_fifo.
/b
On Oct 12, 2009, at 12:55 PM, William King wrote:
I don't know if this was mentioned yet. It would be useful to have a
way
to have the parking lot automatically find the next available spot and
tts it to the person parking the call.
Then the auto unpark would
NO this is in the XML... not in the db table.
/b
On Oct 12, 2009, at 11:16 PM, srinivasula reddy wrote:
in the same way there is any table for groups, how many groups are
there? and information about groups(directory/default.xml this file
having the group configuration).
I wouldn't call it donating per se... Its just giving it a place to
live with easy access for end users without having to do anything
extra go get it! ;)
/b
On Oct 12, 2009, at 11:27 PM, Tihomir Culjaga wrote:
this will be perfect ... but it is up to Yuriy if he is willing to
donate
Its based on the directory and who is in the group... check out the
defaults it does exactly this on the 2000 range if I recall correctly.
/b
On Oct 12, 2009, at 11:32 PM, srinivasula reddy wrote:
OK, then when i call to group number(911) how it will call to all
the registered members in
I don't think mod_dingaling will do a lookup for host: like sofia will
as it doesn't have the code for that last I checked... I could be
wrong but I don't recall it doing that.
/b
On Oct 13, 2009, at 3:02 PM, Mark Campbell-Smith wrote:
I have a hostname set in vars.conf.xml for the
You shouldn't have to make clean usually ... doing so might break your
tree... You can usually get by with make current that will ensure
the critical things are cleaned and built correctly... every now and
they you'll hit a snafu but we'll usually tell you about it.
/b
On Oct 13, 2009, at
Perfect! Thanks.
/b
On Oct 12, 2009, at 1:15 AM, Seven Du wrote:
http://jira.freeswitch.org/browse/MODCODEC-15
Is it ok I assigned to you ?
Thanks.
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
Did you open a jira and attach all the info?
/b
On Oct 12, 2009, at 3:47 AM, Maciej Aniserowicz wrote:
Yes, I confirmed that with Wireshark (filter rtp and ip.src ==
device ip). RTP packets are sent every 20ms.
MAniserowicz
___
We can host this in our SVN if you wish?
/b
On Oct 12, 2009, at 8:31 AM, Georgiewskiy Yuriy wrote:
ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code, but
seems it work, but should be buggy,
to build need libpt 2.6.5 and h323plus cvs version, i test it now on
fs 1.0.4.
Fixed... svn up.
/b
On Oct 12, 2009, at 1:15 AM, Seven Du wrote:
http://jira.freeswitch.org/browse/MODCODEC-15
Is it ok I assigned to you ?
Thanks.
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
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
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.
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http
wrote:
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
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.?!
It happens on invite during a call not on REGISTER... I can 100%
confirm it works so I do not know what the heck you're doing.
/b
On Oct 10, 2009, at 6:02 PM, Juan Backson wrote:
Hi,
I am still stuck in trying to get curl directory variables to show
up in channel even after the user has
Also check to make sure you're returning the correct format as per the
XML curl page.
/b
On Oct 10, 2009, at 6:02 PM, Juan Backson wrote:
Hi,
I am still stuck in trying to get curl directory variables to show
up in channel even after the user has registered.
Is this something that
How much ram do you have and what distro are you running on, 32bit or
64bit?
/b
On Oct 8, 2009, at 9:46 AM, Dome Charoenyost wrote:
When i try
fsctl max_sessions 1
in CLI . FS set it back to 2xx again
Dome C.
___
FreeSWITCH-users mailing
If you're starting a new topic/thread please click new message. If
you click reply, delete the subject and body and change it.. the
thread is now hijacked.
/b
On Oct 8, 2009, at 6:23 PM, Klaus Hochlehnert wrote:
Sorry for hijacking. Didn’t know that the mailing list system finds
out
http://wiki.freeswitch.org/wiki/Mod_xml_curl Is a great start.
/b
On Oct 7, 2009, at 1:43 AM, srinivasula reddy wrote:
Hi,
i want use mod_xml_curl, the xml files also there in my local
system, i dont want to take from any other system,
can any please tell me how to configure
I would suspect its a PEBKAC. I mean if you could register to a
gateway that rejected auth... what purpose would auth serve in the
first place?
/b
On Oct 7, 2009, at 8:48 AM, Nicolas Brenner wrote:
Is there some way to make FS register with the gateway that is
rejecting the
No!
/b
On Oct 7, 2009, at 8:37 AM, Muhammad Shahzad wrote:
Great. Just added it. Is there any user limit on this?
Thank you.
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
Bet you its inband dtmf and you need to start the dtmf detector.
/b
On Oct 7, 2009, at 8:11 AM, Andy wrote:
I can hear the IVR message played down the phone line so outgoing
audio is ok.
___
FreeSWITCH-users mailing list
You'll need to set presence_id so it can work properly. SEE the
default config that does exactly that with sofia_contact in the dial-
string on the domain.
/b
On Oct 7, 2009, at 8:26 AM, Michael Jerris wrote:
When calling the Bridge application with data parameter of sofia/
201 - 300 of 2318 matches
Mail list logo
