I'm using VQManager (there is a 30 day trial) and it's useful for seeing who
does what / when per call; it's very easy to install...
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Frank @
Impact
Sent: Thursday, December
cut
about 31 seconds, no issues at all with either port audio or gtalk,
Could anyone point me to the right direction for the sofia_sip profile
setup?
Your helps are greatly appreciated
Thanks,
Chris
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the settings regarding nat acl and localnet
acl.
Chris
On Fri, Dec 11, 2009 at 11:25 AM, Frank Carmickle fr...@carmickle.comwrote:
On Fri, Dec 11, Chris Chen wrote:
Hi there, I have very strange behaviors for my SIP endpoints with FS SVN
trunk 15905.
Is this a change in behavior
Hi Mike, the fs console log with sip trace on the internal profile is
attached in the pastebin below,
http://pastebin.freeswitch.org/11483
could you please take a look at it?
Thanks,
Chris
On Fri, Dec 11, 2009 at 12:51 PM, Michael Jerris m...@jerris.com wrote:
As i said multiple times on irc
Thanks Mathieu, but I am on SVN r15912 now.
Chris
On Fri, Dec 11, 2009 at 2:09 PM, Mathieu Rene mrene_li...@avgs.ca wrote:
Its not sending to the right Contact: header in the 200 OK packet. This
was fixed in r15870, you have to update.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1
Thanks Brian for your explanation, could we still keep the option to set the
extrip ip, as my DLINK DIR-655 UPNP is not working reliably, and I believe
many other routers have similar issue.
Chris
On Fri, Dec 11, 2009 at 2:45 PM, Brian West br...@freeswitch.org wrote:
You set the extrtp ip
in
September with Brian lamenting that Skype was hard to work with on this.
I know I could use mod_skypiax; but having a native solution would be
one less IT headache.
Thx, Chris.
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One suggestion to you, please never consider the GXW4108 for any business
use unless just in LAB. The GXW4108 will work when it is working,but I can
tell you within one year you will be regretting your choice for use of
GXW4108 if you put into production for business use.
Chris
On Wed, Nov 25
You haven't really put it into production for more than one year. The issue
with GXW4108 is that after some time, say a couple of months, either all FXO
ports not working, or worse, some FXO ports not working, but after power
recycling, they will come back to work for some time until on strike
Your SIP UA needs to take the info in the 401 and use it to digest
authenticate. If you trace a SIP UA that supports authentication you will see
that they also get the 401/407 and only then are able to authenticate. This
is just a fact of how digest auth works in SIP ... see section 22.4 The
you have to define the extension john or bob or whatever number you
want in the dialplan for the context public.
Just follow your jingle profile you define. Simple, no other tricks.
Thanks,
Chris
On Wed, Nov 4, 2009 at 11:56 AM, Jonathan Barou jba...@sqli.com wrote:
Hi everybody,
I
with 3COm 3102 phones myself.
Chris
On Tue, Nov 3, 2009 at 3:03 PM, Dave Stevenson steve...@primrosebank.netwrote:
Tihomir,
thanks for the link, but actually, I had already found/downloaded/read and
almost understood that document !
However, the options to log into the phone and configure
My favorite part of this 'civilized' discussion on IRC was when DelphiWord and
diegoviola sat around tryin to take the piss outta stkn on this issue for
seemingly no reason. Thanks for making the channel a cool place, guys ;)
On October 31, 2009 07:32:03 pm Meftah Tayeb wrote:
Anthony
Do you have a debug level message from switch_ivr_originate.c in your log?
Channel is already up, delaying proxy mode 'till both legs are answered.
Set bypass_media b4 each bridge. It is unsetting on you and setting
bypass_media_after_bridge because you already answered the channel running
the
Depends on your codec. Frame size = ptime. So it can vary from call to call,
but most you will find it to be 20ms.
So if we are using speex-wb at 20ms, each frame is 0.02 seconds long. 50
frames = 1 second of sound.
On October 28, 2009 12:37:56 pm Matthew Fong wrote:
I noticed that the
poor bbhenry :)
Added in r15207 please test and update docu if necessary:
http://wiki.freeswitch.org/wiki/Variable_api_on_answer
On October 23, 2009 11:02:07 am Anthony Minessale wrote:
it's probably related to escaping the data.
I was sick of watching you suffer so i added api_on_answer
one real quick way would be put different GET var in each server's binding
On October 23, 2009 03:46:11 pm Kristian Kielhofner wrote:
Can't you use different contexts or something else to tell them apart?
On Fri, Oct 23, 2009 at 3:34 PM, freeswitch noob
freeswitch.n...@gmail.com wrote:
I
a case of RTFM.
-metik
- Original Message -
From: Metik
To: Metik ; freeswitch-users@lists.freeswitch.org
Sent: Friday, October 23, 2009 5:48 PM
Subject: Re: [Freeswitch-users] Hostname
Please note that this would essentially be taking Chris' suggestion a
little further
If you really wanted: http://php.net/manual/en/book.sqlite.php
But I would recommend you make use of ODBC to use a client/server RDBMS.
Here's some good reading:
http://www.sqlite.org/cvstrac/wiki?p=WhenToUseSqlite
On October 20, 2009 10:53:01 am homqua wrote:
Now I am building a PHP SOAP Web
There's a check_sip plugin for nagios if you're into that sorta thing
On October 19, 2009 12:39:53 pm Michael Collins wrote:
2009/10/19 Christian Löschenkohl christian.loeschenk...@xpirio.com
hello
we have the problem here that our freeswitch server freezes from time
to time (no sip
- and really just exposing a problem I've always had
before?
The config is (50 Polycom Phones - NAT - Internet - Amazon EC2)
I would really appreciate some pointers on what to look for; additional
trace that might reveal something.
Thanks, Chris.
On Fri, 16 Oct 2009 20:06:52 -0700, Chris Fowler ch
?
Thanks, Chris.
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Hi Muhammad, the simple and reliable solution for you where SIP is being
blocked is add
conf+...@conference.freeswitch.orgconf%2b...@conference.freeswitch.orgto
your Goolgetalk buddy list, and you can call from there to join the
conference, simple and straightforward.
Chris
On Wed, Oct 7, 2009
This happens with our polycoms as well ... NAT on phone and PBX. Still haven't
had time to look into it so I disabled the sound for new message waiting ...
for now it doesn't keep beeping every few minutes.
On September 23, 2009 08:08:39 pm Brian West wrote:
NO I have never seen it happen what
Check out the variables ringback and transfer_ringback. The local extension in
the default dialplan is a good example.
For romance, I recommend 80s rock ballads. YMMV.
On September 11, 2009 12:22:20 pm Dome Charoenyost wrote:
Dear Sir,
Is posible to play music for background when
There are a few ways you could go about dropping into a conference and playing
the song in from a separate channel.
On September 11, 2009 01:47:43 pm Dome Charoenyost wrote:
2009/9/12 Chris Burns ch...@cloudtel.com:
Check out the variables ringback and transfer_ringback. The local
extension
with server.xml), for the invite requests to some
FreeSWITCH built-in accounts such as user+bla b...@your jabber server
ext+blah blah@ your jabber server,
they will be automatically accepted.
Hope this helps.
Chris
On Fri, Aug 21, 2009 at 1:30 AM, Tapan Parikh tpar...@gmail.com wrote:
Hi Folks
Thanks!
João Mesquita wrote:
No one said a thing but I really feel like initiatives like this
should be cheered.
Thank you for this app and thank you for making it free as well.
On 8/14/09, Chris Danielson ch...@maxpowersoft.com wrote:
Announcing the release of FreeSWITCH Console
Hi, could you please check the destination number in your dial string? If it
is the right format, one of the reasons could be that number is not in
service
when you get 403 response from the SIP gateway.
Thanks,
Chris
On Mon, Aug 17, 2009 at 8:10 AM, NOx-WHV enno.egb...@googlemail.com wrote
:
http://www.chrisdanielson.com/2009/08/14/release-iphoneipod-touch-freeswitch-console/
iTunes Store Link:
http://itunes.apple.com/WebObjects/MZStore.woa/wa/viewSoftware?id=319105221mt=8
http://itunes.apple.com/WebObjects/MZStore.woa/wa/viewSoftware?id=319105221mt=8
Kind Regards,
Chris Danielson
I couldn't imagine managing a conference without a GUI. I need to see who is
making noise so I can boot/mute em ;)
If I were you I would dive into ESL and make a simple web app to frontend
the conferences. There will surely be something in contrib to get you
started.
On Thu, Aug 13, 2009 at 8:48
private IP address.
Chris
On Wed, Jul 29, 2009 at 12:30 PM, Dale fdh...@gmail.com wrote:
I have a freeswitch server located behind 1:1 nat that I have been
using to test things with. Today I upgraded to the latest trunk and
began having troubles with my private ip being exposed in the sdp
Jingwei, can you show your console log when somebody is calling you from
gtalk client? Will it really hit 888 in your dialplan?
Thanks,
Chris
On Mon, Jul 13, 2009 at 5:27 AM, Jingwei Yang jingwei.y...@gmail.comwrote:
Hi Chris, sorry for the late reply. Have been quite busy last few days.
I
Hi Ray,
This was a problem some time ago (couple of months ago). Are you running the
latest build?
Chris.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Raymond
Chandler
Sent: Wednesday
Jingwei, I don't know if you have the 888 defined in default.xml? also you
have to define $${domain}.
please do dl_debug on from fs_cli, and watch the console logs and see
what's going on when you try calling from external. Most likely your
dialplan is not correctly defined.
Chris
On Mon
Brian, I would like to be one of the volunteers helping to report issues.
Chris
On Fri, Jun 26, 2009 at 11:13 AM, Brian West br...@freeswitch.org wrote:
FreeSWITCHers,
We have written an extensive guide on posting bugs to jira. Over the past
few weeks everyone has been a little lax
setup behind the NAT router.
On Thu, Jun 25, 2009 at 1:31 AM, Jingwei Yang jingwei.y...@gmail.comwrote:
Hi Chris, thanks for your help. Here's my client.xml
include
!-- Client Profile (Original mode) --
!-- to use this profile take the x- away from the open and close tags so
its
...@192.168.0.250 entering state [calling][0]
2009-06-25 12:23:32.162929 [DEBUG] sofia_glue.c:1692
sip:1...@192.168.0.250sip%3a...@192.168.0.250;transport=tcp
Setting proxy route to sofia/internal/1...@192.168.0.250
Chris
On Thu, Jun 25, 2009 at 12:06 PM, Brian West br...@freeswitch.org wrote:
Can you verify
Brian, 13952 fixed the problem, now SIP over TCP to my Exchange 2007 UM is
working.
Thanks for your great work.
Chris
On Thu, Jun 25, 2009 at 12:53 PM, Brian West br...@freeswitch.org wrote:
I found the problem... the fs_path refactor regression number 2 was just
fixed.. It was assuming
you expect, and
check the console log from fs_cli when you do gtalk calling to your gmail
client, you will find out the solution to your issue.
chris
On Thu, Jun 25, 2009 at 10:15 PM, Jingwei Yang jingwei.y...@gmail.comwrote:
Hi Chris. thanks for the reply. Here're my answers.
On Thu, Jun 25
Please provide your client.xml detail with confidential information
crossout, I have gtalk client and server working properly behind the NAT.
I should be able to help you.
Chris
On Wed, Jun 24, 2009 at 11:42 PM, Jingwei Yang jingwei.y...@gmail.comwrote:
Thanks seven. External IPs have sound
?
On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb larc...@yahoo.com wrote:
I’m sorry Chris, but I don’t know where the look for the “global sip.cfg
and mac/phone specific cfg” settings. I also looked for digitmap but could
find nothing.
Can you be more specific?
Thanks, Lars
*From:* freeswitch
Sounds like a config issue in the dialplan/ tag. Check global sip.cfg and
mac/phone specific cfg. When you are dialing on-hook I don't think it will
use your .digitmap or .digitmap.timer settings. When you dial off-hook it
sure will.
On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb larc...@yahoo.com
with dedicated hardware resources and a set
geographic location, then these guys do it. Kind of the best of both
worlds. Just a quick 2 cents...
Regards,
Chris
Brian West wrote:
On May 26, 2009, at 10:11 AM, Kristian Kielhofner wrote:
Hey Brian,
FreeSWITCH in EC2 is a bit of a mystery
);
That should do it.
Regards,
Chris
**
Baskar wrote:
*Hi,
I have an issue in transfer the call through JavaScript session.
In JavaScript Session i have dialed 2 numbers
One is mobile number and another one is extension number
I want both the call to transfer in the conference room using
value=$${outbound_caller_name}/
param name=caller-id-number value=$${outbound_caller_id}/
param name=comfort-noise value=true/
/profile
Thanks, Chris.
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. voicemail)?
There is a uuid_transfer that will allow you to route them accordingly.
Regards,
Chris
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and freeswitch is svn trunk At revision 13154.
Thanks,
Chris
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wrote:
You're missing the REAL reason it failed... the libtool warning wasn't
really what failed. Please do not make -j and do just make and see what
really fails.
/b
On Apr 26, 2009, at 1:45 PM, Chris Chen wrote:
Making all in nua
LTCOMPILE nua.lo
LTCOMPILE
Thanks Mathieu, after installing zlib-devel.x86_64, no eerors any more.
On Sun, Apr 26, 2009 at 3:00 PM, Mathieu Rene mrene_li...@avgs.ca wrote:
You dont have the 64 bit zlib-devel package, install it
On 26-Apr-09, at 2:59 PM, Chris Chen wrote:
Hi Brian, thanks for your quick reply, here
, 2009 08:29
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Using Variables in Dialplans
On Wed, Apr 22, 2009 at 5:20 PM, Chris Fowler ch...@fowler.cc wrote:
Brian, Michael,
Thanks for the help - I had read that but not fully comprehended it until you
spun it the way
something simple...
Thanks, Chris.
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=answer/
action application=sleep data=2000/
action application=voicemail data=default ${domain_name} 2001/
anti-action application=voicemail data=default ${domain_name}
2001/
/condition
/extension
Thx, Chris.
From: freeswitch-users-boun...@lists.freeswitch.org
] switch_core_session.c:1020
switch_core_session_thre
ad() Close Channel sofia/internal/1...@**.***.219.221 [CS_DONE]
Thank you for the help.
~Alex
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Chris
. This is to be able
to hangup
; a call in the case of a phone disappearing
from the net,
; like a powerloss or grandma tripping over
a cable.
This works with one of my ITSP as they provide SIP trunking via *
Hope this helps.
Chris
-PROCESS cmd=set data=default_provider_username=/
X-PRE-PROCESS cmd=set data=default_provider_password=secret/
X-PRE-PROCESS cmd=set
data=default_provider_from_domain=sip.flowroute.com/
Chris.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
from a gateway that does not exist.
What's the output of sofia status (F5 on the console)? It should show:
sip.flowroute.com gateway sip:...@sip.flowroute.com REGED
Chris.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch
doing something incredibly silly.
myConference dialplan is really nothing more than: action
application=conference data=50...@default/
Love this application!
Regards,
Chris
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value=true/
The phones connect on port 5060 - nothing specical to config in the
mac-phone.cfg file for the phone; just host, port, user/pass.
Chris.
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-PROCESS cmd=set data=external_sip_ip=insert EIP here/
Re: Wiki
Yup I need to get on this.
FWIW - I work for RightScale; our computer room is empty except for
routers and switches. *Everything* else lives in the Cloud :-)
Cheers, Chris
Hi Brian, looks like this Evil is calling everywhere today on port 5060,
please see my asterisk log
[Apr 3 11:13:42] NOTICE[16617] chan_sip.c: Failed to authenticate user
MeucciSolutions
sip:meuccisoluti...@66.96.218.5sip%3ameuccisoluti...@66.96.218.5
;tag=as05dbf888
[Apr 3 11:25:12]
It is strange this IP is from US
66.96.218.5USUNITED STATESPENNSYLVANIASCRANTONNETWORK OPERATIONS CENTER INC
On Fri, Apr 3, 2009 at 1:53 PM, Gabriel Kuri gk...@ieee.org wrote:
I heard about this a few days ago, they claim it's not them, but someone
trying to harm their reputation ...
Try turning off comfort noise completely in the conference profile? My 650s
sound great in conference w/ PCMU and G722
On April 1, 2009 03:10:35 pm Giovanni Maruzzelli wrote:
To make a long story short, a ground loop is when an electric circuit
is made between different audio device that are
was the elimination
of the 250ms sleeps; and the change to:
macro name=welcome pause=250
I'm running build 12782; should this have fixed it? If so, I will
follow the bug reporting instructions you sent earlier.
Thanks, Chris.
Here's the errors caught today on my production system.
2009-03-27 07:20:41
can capture to assist?
Thanks, Chris.
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Any thoughts on why FS saw all digits 1029 but only reports '029'?
2009-03-25 10:48:45 [DEBUG] switch_ivr_menu.c:364 play_and_collect()
digits '029'
Config:
menu name=main_ivr
greet-long=phrase:welcome
greet-short=phrase:top-menu
First off what SVN rev? Remember when reporting issues try to include all
the information you can!
Oops; forgot that - FreeSWITCH Version 1.0.trunk (12647)
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Thanks for the tip Brian.
Here's a link to the valgrind output : http://cfowl.postinbox.com/vg.log
Chris.
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. Currently running version FreeSWITCH Version
1.0.trunk (12604).
This is seen both when FS is being used (~200 calls/day, and over the
weekend when ~5 calls/day).
How can I best debug this?
Thanks, Chris.
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Jay : what happens in your dialplan ?
Nothing special; no external script execution just default pattern
matching to route to extensions (per the stock config).
Brian: Can you update to SVN trunk as of now?
Yup, I will pull the trunk and report back in 24 hours.
Chris
Brian: Can you update to SVN trunk as of now?
I updated - version reports: FreeSWITCH Version 1.0.trunk (12631)
Only difference I note with this build is that upon shutdown FS now
SegFaults. The mem/cpu usage continues to slowly climb.
snip
2009-03-16 20:59:32 [CONSOLE]
and
sessions initialized and running in memory? (I'm looking to clone and
serialize them and store them on a separate server).
2) Any helper methods or ways that I can re-construct the channels and
sessions into memory on a freshly started instance of FreeSWITCH?
Kind Regards,
Chris Danielson
Michael,
Thanks! I'm joining up now.
Regards,
Chris
Michael Collins wrote:
On Wed, Mar 11, 2009 at 8:25 AM, Chris Danielson ch...@maxpowersoft.com wrote:
Hello Guys,
I have a question regarding how I can do the following within the FS C API.
The devs love it when people get down
apt-get install unixodbc-dev
On March 5, 2009 11:02:45 pm mashudi wrote:
Hi Folk,
i got error while conduct ./make freeswitch-snapshot on debian 2.6 x86
here is the error :
/usr/bin/ld: cannot find -lodbc
collect2: ld returned 1 exit status
make[2]: *** [libfreeswitch.la] Error 1
Making
I'm having a problem that calls to the auto-attendant won't transfer. I know
this has been a problem in the past but thought that it was fixed. Whenever
I enter an extension (or press a key to transfer me to one), the call just
hangs up. I ran Freeswitch on the console, but all I see happening is
Hi all, I¹m just starting playing around with FS and I¹ve searched for the
answer to what I think is an easy question but I can¹t find it. I have FS
running, 2 X-lite clients on 2 different computers connected using the
preconfigured 1000 and 1001 extenstions. Both can call each other and
to see
the sofia status output.
/b
On Feb 11, 2009, at 2:32 PM, Chris Elam wrote:
$cmd = api originate sofia/mydomain.com/1...@192.168.15.50
bridge(sofia/mydomain.com/1...@192.168.15.50);
The result I get is : -ERR DESTINATION_OUT_OF_ORDER
That's it, worked perfectly, thanks a bunch!
On 2/11/09 3:59 PM, Brian West br...@freeswitch.org wrote:
try sofia/myinsideip/1000 and sofia/myinsideip/1001
I sure hope it doesn't say myinsideip on there and you only tried to
hide your IP.
/b
On Feb 11, 2009, at 2:54 PM, Chris Elam
Closing the connection will force the server to close any open transactions, as
well as release recordsets in local memory that reference the connection.
However curs is not a recordset. An SQL update is going to return an integer
(rows affected) or boolean depending on the which server you
Great thanks to Jason for sharing Cherebrum's great discovery, this works
like a charm on my Ploycom IP 320 with G722 codec.
Chris
On Tue, Jan 27, 2009 at 4:48 PM, Jason Garland jgarl...@jasongarland.comwrote:
Something like this might do it... ;)
?xml version=1.0 standalone=yes?
!-- SIP
.
Again, patches are welcome. :)
-Chris
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Fidibus,
Make sure that you also have installed unixodbc. As shown here:
http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc#unixodbc
Kind Regards,
Chris
fidibus83 wrote:
Oh, I'm sorry.
Should I comment mod_spidermonkey_odbc in
root/freeswitch/build/modules.conf again
Anytime. Good luck.
fidibus83 wrote:
Sorry Chris, my mistake. Mod_spidermonkey_odbc.so does exist now.
I wanted to said that I'm sorry for my stupid mistake that I uncomment
the wrong thing.
But thank you very much for your help
Make sure to follow this example:
http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc#General_Configuration
Remember to use the isql client to test your DSN connection. When that
passes you'll be home free.
Regards,
Chris
Stephen Crosby wrote:
I'm assuming you have your driver set up
: +
+ +
+ make install +
+--+
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
Thanks
Chris
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I'm no expert, but I believe in media bypass mode freeswitch isn't handling
media so it's not a fs fix, it would be the quality of connection for each of
the originator/terminator, fs just directs each endpoint to set's up a point to
point connection for RTP.
Is this right?
mszla...@aol.com
If you need to do load balancing, you could set up a conference_a domain on one
switch, conference_b on the second, conference_c on the third, then use
xml_curl to dialplan and bridge the call to the right domain...
But again, I am no expert... Just a noob trying to be creative. :P
Chris
How do I get off this mailing list? I am not a programmer and have no idea
why I'm signed up for this mailing list. If someone can help, please email
me directly and let me know what to do.
Thank you.
[EMAIL PROTECTED]
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language model. %s\n, lm);
goto end;
}
*Chris Danielson*
Software Consultant and Co-Founder
Web: MaxPowerSoft, LLC http://www.maxpowersoft.com/
Index: mod_pocketsphinx.c
===
--- mod_pocketsphinx.c (revision 9003
.
Attached is my svn diff.
Kind Regards,
Chris
--
*Chris Danielson*
Software Consultant and Co-Founder
Web: MaxPowerSoft, LLC http://www.maxpowersoft.com/
Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Index: mod_conference.c
This patch is the equivalent of automatically running the following
command every time a new conference starts.
conference conference name record my auto record
location/conference name
*Chris Danielson*
Software Consultant and Co-Founder
Web: MaxPowerSoft, LLC http://www.maxpowersoft.com
http://jira.freeswitch.org/browse/MODAPP-112
Thanks!
Chris
*Chris Danielson*
Software Consultant and Co-Founder
Web: MaxPowerSoft, LLC http://www.maxpowersoft.com/
Michael Jerris wrote:
Can you please attach this patch as a file to a bug on
jira.freeswitch.org?
Mike
On Jul 7, 2008, at 5:36
the switch_channel_expand_variables call.
Kind Regards,
Chris
*Chris Danielson*
Software Consultant and Co-Founder
Web: MaxPowerSoft, LLC http://www.maxpowersoft.com/
Brian West wrote:
Dan and I had a chat about this issue. I had to think on it but I
think Dan and I have agreed on a solution.
param name
Added this information to:
http://wiki.freeswitch.org/wiki/Channel_Variables
Michael S Collins wrote:
Brian,
This scenario is right up my alley so I will be happy to update the
wiki.
-MC
Sent from my iPhone
On Jun 6, 2008, at 7:55 AM, Brian West [EMAIL PROTECTED] wrote:
Internal
definitely make my life easier in this
scenario.
Thanks!
Chris
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codec list, and
tried Xlite 3.0 and Bria 2.2 with camera.
Thanks,
Chris
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the error
[EMAIL PROTECTED] not found
Could you guys share the correct syntax for doing SIP to Jingle calls?
Thanks,
Chris
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Updated the wiki.
Regards,
Chris
Leonardo Alves wrote:
Thanks,
Now it is working just fine.
And I really loved the make current to update to the last version.
Leonardo Alves
*From:* Anthony Minessale mailto:[EMAIL PROTECTED]
*Sent:* Tuesday, April 15, 2008 3:45 PM
*To:* freeswitch-users
bit rate? Anyways, the console will let you know if there is a
mismatch and that FreeSWITCH is having to pull overtime in order to
process the sounds.
Cheers,
Chris
Jonas Gauffin wrote:
Hello
I'm launching a javascript that originates a call to a user and then
records a file.
The problem
profile default.xml?
Thanks
Chris
On Thu, Apr 10, 2008 at 2:18 PM, Michael Jerris [EMAIL PROTECTED] wrote:
This smells of a nat related issue. Can you explain the network
layout and where nat is involved.
Mike
On Apr 10, 2008, at 1:32 PM, Chris Chen wrote:
I am new to free switch, and I
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