On Sun, Dec 27, 2009 at 5:01 AM, Karl J. Vesterling k...@ken-ton.com wrote:
Setting the codec negotiation to scrooge resolved my problems w/
CallCentric.
I'd bet that'd do it for him as well.
*Lessons Learned by me:*
1.) Listen to Brian.
2.) When in doubt, refer to rule 1.
Can I get
Lars,
Since this question has come up a few times I'm going to write up a nice
wiki article on it explaining the differences between letting someone in via
an ACL and actually doing digest authentication. In a nutshell, though, it's
this: if the user does digest authentication (with the whole
On Tue, Dec 22, 2009 at 7:58 PM, Joseph L. Casale jcas...@activenetwerx.com
wrote:
Am I correct in presuming that Freeswitch will answer a fax from a local
zap based user
just like it does from an FXO port connected to a POTS line? What I hope
to do here is
catch any call made from that
On Wed, Dec 23, 2009 at 7:39 AM, Fred-145 codecompl...@free.fr wrote:
More information: I can dial the default extensions like just fine.
It's
only when I call any of the IP phones (1001,1002,1003) that the call is
immediately forwarded to the callee's voice-mail when the phone goes off
Hello all!
Because the holidays fall on consecutive Fridays this year we decided to
have a single conference call on Wednesday Dec 30th at the usual time of
11AM CST. The agenda is posted here:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_30
Thanks for supporting the weekly calls. Don't
On Tue, Dec 22, 2009 at 10:50 AM, Larry Marshall l...@marshap.com wrote:
I have set up a second FreeSWITCH box on the same LAN. I have v16018
installed on it and have changed nothing.
I configured a Polycom phone to register one of its four lines to this
second box, but it does not
On Tue, Dec 22, 2009 at 11:35 AM, Yehavi Bourvine yehavi.bourv...@gmail.com
wrote:
Try tracing the calls from both sides with TCPDUMP or enable siptrace on
FreeSwitch. I guess this will give you some clue.
__Yehavi:
Additionally, turn on debugging on the console
It's upgrade-and-test time! The new release announcement is on the main
FreeSWITCH page:
http://www.freeswitch.org/node/224
Please update, test, and report back bugs and questions.
Thanks!
-Michael
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On Tue, Dec 22, 2009 at 11:46 AM, Lars Zeb larc...@yahoo.com wrote:
Yes, the internal profile exists.
Name Type
Data State
=
internal
On Mon, Dec 21, 2009 at 8:07 AM, Andrew Thompson and...@hijacked.us wrote:
On Sun, Dec 20, 2009 at 11:03:46PM -0800, ram wrote:
Hi
its good to hear
any compare document between Vicidial and this project
No document, but briefly:
* More focused on inbound than on outbound (at
On Sun, Dec 20, 2009 at 10:31 PM, Edmar Cruz darklio...@yahoo.com wrote:
Where should I write this line
extension name=international dialing
condition field=${toll_allow} expression=international
anti-action application=playfile
data=misc/you-are-not-authorized.wav/
anti-action
On Sun, Dec 20, 2009 at 6:39 PM, Edmar Cruz darklio...@yahoo.com wrote:
Hi Sir,
How can I allow international calling in the dialing plan but for
select accounts only?
For example i want to restrict 855 to call this ip address
182.138.252.12 using the default configuration..
Hello everyone!
Today's agenda is listed here:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14
Also, we are going to be giving away goodies on some of the upcoming
conferences, so call in and see what we've got in store. :)
For the first 15 minutes we'll let everyone mingle and then we'll
On Fri, Dec 18, 2009 at 2:21 AM, Robin Vleij vi...@fx-services.com wrote:
Hi guys (and girls)!
I'm working on a little bit of ENUM trickery and I tried doing some
(illegal) nested conditions. :-)
What I want to do is to first check enum with the ENUM application,
then depending on the
On Thu, Dec 17, 2009 at 3:59 PM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
Thanks for the hint!
force_transfer_context and force_transfer_dialplan.
I've updated the wiki (I'll add an example once I test it).
I love it when users go all Chuck Norris and Rambo in answering
On Thu, Dec 17, 2009 at 4:01 PM, Frank @ Impact fr...@impactfax.com wrote:
I bit off topic but…
Using FS to send calls sip to the LD carrier.
Some calls have problems where they drop the call or audio drops or
whatever.
The carrier’s first response is that we dropped the call. But
On Wed, Dec 16, 2009 at 4:37 AM, Costa Zikalala costa.zikal...@gmail.comwrote:
Hi All
I understand that to connect to a SIP Provider you have to (amongst other
things) define a Gateway on the External Profile.
But some gateways may be defined on the Internal Profile. What kind of
gateways
On Wed, Dec 16, 2009 at 10:27 AM, RR ranjt...@gmail.com wrote:
Lost that screen that showed me what rev was downloaded but whatever you
get after doing a “svn up” and “make current”. I had done a rm –rf in the
/usr/src/freeswitch directory and then did svn up. Should I have done svn
co
Try setting absolute_codec_string in the dialplan prior to the bridge:
action application=export data=nolocal:absolute_codec_string=G722/
Let us know if that does the trick.
-MC
On Wed, Dec 16, 2009 at 11:13 AM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
Hello everyone,
On Wed, Dec 16, 2009 at 11:30 AM, RR ranjt...@gmail.com wrote:
MC, haha I’m not sure. I think this had happened to me before as well and
nuking the fs dir and then an svn up had fixed it.
I think I’ll just do an svn co and get on with it. Sorry had been following
FS when it first started
And we shouldn't be using 1.0.4 anyway, should we? ;)
-MC
On Wed, Dec 16, 2009 at 3:26 PM, Moises Silva moises.si...@gmail.comwrote:
I've been using FreeSWITCH on Windows lately and seems to work pretty well.
Sangoma has been testing more and more lately the Windows drivers with
FreeSWITCH,
On Tue, Dec 15, 2009 at 8:04 AM, Steve Steffler stevesteff...@shaw.cawrote:
Hi all,
What is the difference between the mod_voicemail vm_message_ext parameter
and the file-extension parameter?
vm_message_ext is a channel variable:
Hello friends,
Just to let you know, I have posted the FS weekly conf call agenda:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_14
It's pretty clean at this point so if you've got things that you'd like to
discuss with the group then please add your items to the list. If you have
items
On Tue, Dec 15, 2009 at 12:11 PM, bcxml bc...@hotmail.com wrote:
I have Freeswitch and Microsoft Speech Server 2007 on the same box
When Speech Server initiates a call, I get a sip error message 480
Here is the internal profile trace...
freeswi...@hd-t2253cn
freeswi...@hd-t2253cn recv
On Tue, Dec 15, 2009 at 2:09 PM, Michael Jerris m...@jerris.com wrote:
Try turning on debug logs, but from this it looks like its not matching any
extensions.
Agreed. console loglevel debug at the fs cli and then make a test call,
capture output, drop into pastebin.freeswitch.org, and post
Can you turn on debug and sip trace and pastebin the console output? Reply
to this thread with the pastebin URL...
I'm sure some of the networking gurus can help.
-MC
On Tue, Dec 15, 2009 at 2:00 PM, Ahmed Naji a.alalo...@gmail.com wrote:
People,
I have a very simple call scenario where calls
Profile is a collection of preferences uses by conferences etc.
In the case of SIP a profile is also the name for the resulting SIP UA
created by a particular profile.
Context is a narrowed down view of something, in the case of the dialplan a
context is a set of extensions. It's like having a
On Mon, Dec 14, 2009 at 9:12 AM, Fred-145 codecompl...@free.fr wrote:
Thanks Anthony for the tip.
Would you say this is a correct representation of things?
Contexts are a set of extensions in conf/dialplan/ (eg. default, public,
etc.)
Extensions are configured through files in
FYI,
The latest pre-release is now available. Usual information is available
here: http://www.freeswitch.org/node/222
Please update as soon as you can. (SVN trunk users do the make current
thing please.) We need your testing and feedback please!
Many thanks for continuing to support FreeSWITCH.
On Mon, Dec 14, 2009 at 11:49 AM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
Hello everyone,
I've been looking for a FreeSWITCH Nagios plugin. Ideally I'd like
to connect to the event socket and run some api commands and return
them (as opposed to checking SIP, for example).
On Sun, Dec 13, 2009 at 5:27 PM, David V. Fansler
dfans...@dv-fansler.comwrote:
Final Count was just over 900 files. At the moment I am putting them in
a logical order – as best I can tell with my limited experience – combining
chapters and providing links from the table of contents to
On Sun, Dec 13, 2009 at 9:10 PM, Thangappan.M thangappan...@gmail.comwrote:
I've seen the source code of mod_rss and mod_voicemail. I was not able to
get it.
What are the steps do I need to follow for implementing IVR using IVR
library?
Is there any documentations available for knowing
On Fri, Dec 4, 2009 at 7:58 AM, Neil Patel ne...@cs.stanford.edu wrote:
Hi All,
I haven't found a substantial example of IVR applications implemented in
lua. Can anyone suggest where to look? My issue has to do with appropriate
coding style.
I am implementing a voice message board
On Mon, Dec 14, 2009 at 11:47 AM, Dmitry Bely dmitry.b...@gmail.com wrote:
I'm playing with demo IVR from FreeSwitch distribution and have a
problem with language settings. I would like to use Russian as a
default language for voice messages so I set in vars.xml
X-PRE-PROCESS cmd=set
Come one, come all!
http://bit.ly/8KzHCZ
Talk to you soon!
-Michael
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On Fri, Dec 11, 2009 at 3:11 PM, bcxml bc...@hotmail.com wrote:
I am very new to Freeswitch so please accept my appologies if these
questions
seem to be trivial
I am trying to setup FreeSwitch to work with Microsoft Speech Serevr 2007.
I
have been successful in getting Freeswitch to pass
On Thu, Dec 10, 2009 at 2:13 AM, Julian Lyndon-Smith aster...@dotr.comwrote:
Sometime next week I hopefully am going to start a document that
follows my progress in setting up a FS system from scratch, with all
the pitfalls and successes. A kinds of warts and all story.
Alongside this blog
On Thu, Dec 10, 2009 at 3:40 AM, Fred-145 codecompl...@free.fr wrote:
No publisher, although uploading and selling books (deadtree or online) is
easy with companies like www.lulu.com
I was just thinking of some way to learn FS gradually and effectively. The
frequent problem with wiki's, is
On Thu, Dec 10, 2009 at 8:42 AM, Otis ab...@greatiam.com wrote:
I have 2 FS servers FS1 (aka medion) and FS3 (callweaver). These are set
as gateways and register with each other. I wanted all users on FS1 to
dial those on FS3 with prefix 33 ie extn 1001 on FS3 will be dialed as
331001 on
On Thu, Dec 10, 2009 at 9:07 AM, Alberto Escudero aep.li...@it46.se wrote:
Hi,
I am currently creating IVR using the functions provided in the XML
dialplan
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ivr
Using functions like this
entry action=menu-play-sound digits=1
On Thu, Dec 10, 2009 at 10:17 AM, Brian West br...@freeswitch.org wrote:
Use the BKW method... three to four word sentences to describe what to
do... its very poetic! Or is that haiku?
/b
Update to latest
Did you type make current yet?
Tony hates build skew
-MC
On Thu, Dec 10, 2009 at 1:11 PM, David Knell d...@3c.co.uk wrote:
On Thu, 2009-12-10 at 10:26 -0800, Michael Collins wrote:
Update to latest
Did you type make current yet?
Tony hates build skew
Brilliant.
Michael Collins-san
Shrinks all usual advice
Into one Haiku.
--Dave
I
On Thu, Dec 10, 2009 at 2:16 PM, Julian Lyndon-Smith aster...@dotr.comwrote:
Ok. The journey begins.
http://makingfs.blogspot.com/
Don't know if you want to add this link to the website or wiki.
Julian
Excellent work! Thanks,
-MC
Asterisk deadlocked
Why does it suck so badly?
Use
FYI,
Here's the agenda for tomorrow's conference call:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_11
Please be ready to join at 11AM CST! :) Don't forget to bring your agenda
items, questions, and a willingness to help out with our various janitor
projects.
Thanks!
-MC
On Tue, Dec 8, 2009 at 3:59 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Thought I’d send this little hurrah! As there seems to have been a lot
of negativity on this list lately.
I hereby multiply all the negative comments by -1. :P
-MC
I found the rosetta stone useful though woefully lacking in volume.
I guess that's true overall with the project.
Documentation is neither easy nor glamorous. The woefully lacking
documentation has been provided by a little group of people who've done a
big bit of documenting and a big group
On Tue, Dec 8, 2009 at 3:46 AM, Jon Bruel j...@consiglia.dk wrote:
I got the combination Lua with direct access to the core Sqlite database
to work. Hurray, maybe I’m not as stupid as A.M II hints…
Tsk tsk! He didn't actually hint that you were stupid - all he said was
that doing ODBC and
Greetings,
The FreeSWITCH developers have uploaded the latest and greatest FreeSWITCH
1.0.5 pre-release version. Please check out the release
announcementhttp://www.freeswitch.org/node/220.
Let's all get updated as soon as possible. Also, please report bugs right
away and follow up when the
On Sun, Dec 6, 2009 at 11:30 PM, Samuel Abekah-Mensah ab...@greatiam.comwrote:
Pardon me if this has been addressed already.
How does one go about having in the simplest instance 2 servers
registering with each other on startup whereby the users registering
would be able to call each other.
Forgive me if I ask the obvious questions...
Did you make in src/libs/esl before doing make phpmod ?
Did you install the php-devel stuff?
-MC
On Mon, Dec 7, 2009 at 9:27 AM, Kendall Stauffer k...@ksac.com wrote:
I have downloaded and compiled freeswitch, and it runs fine, can
compile
On Mon, Dec 7, 2009 at 10:11 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi!
How can I access the variables that are defined in a users xml file?
For example, say user 1000 has a variable called smsnumber, as defined
below:
include
user id=1000 mailbox=1000
params
to do and that was the first thing I tried,
but nothing is passed to the script.
In a case like this, what defines if variable smsnumber is taken from
the A path or B path? (The A path does not have smsnumber defined)
On Tue, Dec 8, 2009 at 5:25 AM, Michael Collins m...@freeswitch.org
wrote
On Mon, Dec 7, 2009 at 11:09 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Perfect...
action application=set_user data=${dialed_extension}@
${domain}/
works like a charm.
Another satisfied customer! :P
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On Mon, Dec 7, 2009 at 2:02 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Hi
Is it possible to trap on DTMF on a bridged call within an LUA script?
I’ve tried setting the gateway to use inband, but no joy. It looks like I
could use start_dtmf, but I can’t see how to launch
FYI,
The agenda is here:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_04
Please call in! :)
-Michael
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Looks good so far. Try oz list and oz dump 1 and see what happens.
-MC
On Thu, Dec 3, 2009 at 10:36 PM, Neil Patel ne...@cs.stanford.edu wrote:
Thanks all for your help. I got around this by running ./Setup and
installing wanpipe in TDM API mode (it says it's the default for FS). I then
On Thu, Dec 3, 2009 at 9:46 AM, Samuel Abekah-Mensah ab...@greatiam.comwrote:
I have copied 1001.xml in directory/default to a test user 2319.xm
changing or instances of 1001 in the file to 2319. I then went into
default.xml in directory folder and in one of the groups just mimicked
1001
On Thu, Dec 3, 2009 at 10:34 AM, Samuel Abekah-Mensah ab...@greatiam.comwrote:
Hi
Sorry .xm is a typo. I actually shut down the server and restarted. The
log says I need to create a domain of aaa.bbb.ccc.ddd (which is the
server IP address ) and then put the user in that domain. Isn't the
On Thu, Dec 3, 2009 at 10:29 AM, David Laperle dlape...@rsslex.com wrote:
Hi guys,
i have a weird problem with my dialplans. For the moment, i have only 2
«usable» extensions. They were working #1 yesterday, but this morning i
realize i forgot to compile mod_python, so i go back into my
2009/12/2 João Mesquita jmesqu...@freeswitch.org
What MC meant was mod_xml_cdr, not mod_xml_curl. Just to avoid confusions.
JM
Thanks for catching my typo! :)
-MC
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On Wed, Dec 2, 2009 at 6:47 AM, eaf erandr-j...@usa.net wrote:
What would be the best way of making originate() run call through a dial
plan
(compared to directly going to a specified VOIP gateway). Would it be
loopbacks, i.e. smth like this?
/opt/freeswitch/bin/fs_cli -x originate
On Wed, Dec 2, 2009 at 8:39 AM, John Platts john_pla...@hotmail.com wrote:
I have uploaded the dialplan and JavaScript files used to process calls to
MODENDP-272. I have even done a make current to revision 15755, and the
blind transfer is still failing.
John,
Thanks for keeping the guys
On Wed, Dec 2, 2009 at 9:21 AM, Artem Shiyanov shiya...@gmail.com wrote:
1 - config
2 - I've done this with programming
3 - suppose programming would be needed
Just to clarify, when you say programming there are different levels of
involvement. For example, you can do programming in C which
On Wed, Dec 2, 2009 at 9:43 AM, François Legal de...@thom.fr.eu.org wrote:
No, my voicemail extension (I have 2 actually) is called
vmain_unregistered_user, so in voicemail.conf.xml I have :
param name=vmain_extension value=vmain_unregistered_user/
But still (and I don't even know if I'm
On Wed, Dec 2, 2009 at 9:58 AM, Frank Carmickle fr...@carmickle.com wrote:
On Wed, Dec 02, Otis wrote:
Snip...
Thanks.
I would like all extensions on say server A to be contactable by those
on server B and vice versa.
The example I gave you should get you started. Let us know how
This seems like an interesting niche project. I think that if you have
programming skills then the community can provide the PBX/VoIP knowledge to
help you get over the hump. I would recommend that you write up a document
describing all the features that this module would need to provide. Reply to
I love mod_xml_cdr :)
My sentiments as well.
-MC
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On Wed, Dec 2, 2009 at 10:04 AM, Martin Rodriguez gmartin...@gmail.comwrote:
Hi list;
I'm new to FreeSWITCH, I'm working with for 6 years with Asterisk and
10 years in VoIP (Cisco). I need a reference guide to start working
with
FreeSWITCH. I download the official documentation, it would
On Wed, Dec 2, 2009 at 12:08 PM, Erwin Davis davis.er...@gmail.com wrote:
Hi, Anthony,
Thanks for your reply.
When I type the command below, I got the error,
Unknown target hd-sound-install
make[1]: *** [hd-sound-install] Error 1
make: *** [hd-sound-install] Error 2
I found out that
On Sun, Nov 29, 2009 at 10:06 AM, Puskás Zsolt erro...@gmail.com wrote:
Hi Guys!
I'm using the latest svn (15711) with the default xml config. Only modified
cdr_csv.conf.xml the line param name=legs value=a/ to param
name=legs
value=ab/
Here is what i do:
1. 1000 calls 1001 (1001
On Wed, Nov 25, 2009 at 9:42 PM, Joseph L. Casale jcas...@activenetwerx.com
wrote:
I need to make faxing easy for some very computer illiterate folk. I am
using an email
service and going to use procmail to print anything incoming automatically
but they cant
get the hang of scanning to an
I'm not entirely sure that I understand your question, so I am going to ask
a few questions to clarify.
Are you looking to have analog telephones receive incoming calls, like in a
call center? Is that why the user of the analog phone would need to log in
and log out?
I would recommend checkout
On Tue, Dec 1, 2009 at 2:09 PM, Joseph L. Casale
jcas...@activenetwerx.comwrote:
In this case, the $1 will only contain whatever is in the parens in your
expression, i.e.
condition field=destination_number expression=^(\d+)$
What do you have for your expression?
-MC
Well, untested of
Hi folks,
I'm doing a little survey to get an idea of what everyone prefers to use for
their operating environment, like 32 vs. 64 bit, Linux vs. Windows, etc.
Please log in to the main page and check out this node:
http://www.freeswitch.org/node/206
Select the environment that you use the most
On Sun, Nov 22, 2009 at 6:09 AM, David V. Fansler
dfans...@dv-fansler.comwrote:
After the help of a couple of people from this list, I now have FreeSWITCH
running - yeah! I have installed X-Lite on a couple of computers and they
dial each other, play music on hold, etc. I have not yet
On Mon, Nov 23, 2009 at 3:25 AM, lakshmanan ganapathy
lakindi...@gmail.comwrote:
Hi,
I'm using perl ESL to control the call in freeswitch.
I'm having the following scenario, but not able to get it right.
Dialplan:
extension name=outbound_soc
condition field=destination_number
On Mon, Nov 23, 2009 at 11:24 AM, Phillip Jones pjinthe...@gmail.comwrote:
Hi there,
I have created a simple conference that works great. The only problem is,
when a participant press # it exits the call. So when a user enters a
conference with a PIN, and by habit they enter 12345 followed
On Mon, Nov 23, 2009 at 12:17 PM, Phillip Jones pjinthe...@gmail.comwrote:
Thanks for replying.
Well in the log I see:
2009-11-23 15:13:22.015625 [DEBUG] switch_rtp.c:2282 RTP RECV DTMF #:760
2009-11-23 15:13:22.062500 [DEBUG] mod_conference.c:2379 Channel leaving
conference, cause: NONE
On Thu, Oct 15, 2009 at 3:44 AM, god.nirvana god.nirv...@gmail.com wrote:
hi all
how can i get the digits when users in the conference??
and,in conference.conf.xml
control action=mute digits=0/ the action will set another
value?e.g:transfer?
thanks
I'm not sure I
On Mon, Nov 23, 2009 at 4:43 PM, Adam Ford li...@redbonez.net wrote:
I actually tried that, as a guess, based on the configuration output of
fifo list. However I am running a tarball release of 1.0.4, which would
explain why it did not work for me.
I appreciate the feedback, and will
On Mon, Nov 23, 2009 at 4:58 PM, Seven Du dujinf...@gmail.com wrote:
And because it's static string for on-hook members, it's hard to set
dynamically. For now, I'm using a callback way - whenever the sip client
answered the call, it fetch the real connected number from a http server.
That's
On Mon, Nov 23, 2009 at 5:12 PM, Brian West br...@freeswitch.org wrote:
You do realize that the whole concept is OLD skewl. You should be
popping this info via external resources when the agent is bridged to
the caller and the info is there before they are done saying thanks
for calling
On Sat, Nov 21, 2009 at 4:10 AM, Dave Stevenson
steve...@primrosebank.netwrote:
Sorry for the extended forum thread on this subject - This really IS the
last post !
I have now got the ATA to work without the dialplan fix provided by
Michael.
After I'd implemented the fix, I had more of an
On Fri, Nov 20, 2009 at 1:16 AM, Gaurav Singh gaurav...@yahoo.com wrote:
Hi,
Does freeswitch support transcoding between broadvoice (BV32 ) and G711 ?
Try latest trunk. There was a new update just added very recently...
-MC
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On Fri, Nov 20, 2009 at 9:51 AM, Costa Zikalala costa.zikal...@gmail.comwrote:
I've been trying to make phpmod without any success. I've even tried to
./configure --with-php and it did't help.
I've just upgraded to latest svn and am running FS on FC11.
I keep getting this error message:
On Thu, Nov 19, 2009 at 3:33 AM, Dave Stevenson
steve...@primrosebank.netwrote:
Hi,
Can someone please help me understand a little more about Group
configuration ?
I believe that Group Membership is configured in the
\conf\directory\default.xml file
I've done this and the caller groups
On Thu, Nov 19, 2009 at 12:18 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
set enable-3pcc to proxy instead of true
FYI, the wiki entry is here:
http://wiki.freeswitch.org/wiki/Sofia.conf.xml#enable-3pcc
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On Thu, Nov 19, 2009 at 11:46 AM, Artem Shiyanov shiya...@gmail.com wrote:
Hi there!
I've got annoying FS behavior:
There are 2 channels executing the same Java application (application
itself is an IVR). If I try to bridge them with uuid_bridged then both
channels are killed. Here is a log
On Wed, Nov 18, 2009 at 6:54 PM, John Platts john_pla...@hotmail.comwrote:
I have installed FreeSWITCH on our server, and need some help configuring
our FreeSWITCH instance. All of the numbers associated with our FreeSWITCH
instance are in the format: 1NPANXX (where NPA is the area code,
On Thu, Nov 19, 2009 at 12:49 PM, Brian West br...@freeswitch.org wrote:
We have removed the two modules using the reference code from
BroadVoice and added a lib with a new interface from Steve Underwood
and mod_bv.c using this lib... We know their is ONE last bug to be
fixed in the lib
On Wed, Nov 18, 2009 at 8:32 AM, mayamatakeshi mayamatake...@gmail.comwrote:
About mod_fifo, it would be safe to use it in multi-tenancy scenarios where
domains are created and deleted all the time and in consequence, fifos are
created all the time? I mean, fifos are eventually destroyed by
On Wed, Nov 18, 2009 at 1:36 PM, Tim Uckun timuc...@gmail.com wrote:
On Thu, Nov 19, 2009 at 10:04 AM, David Knell d...@3c.co.uk wrote:
Hi Tim,
Here you go:
http://old.nabble.com/echo-cancellation-on-PRI-cards-td22552605.html
Thanks. That's almost exactly the same situation as the one
On Wed, Nov 18, 2009 at 5:09 PM, mayamatakeshi mayamatake...@gmail.comwrote:
On Thu, Nov 19, 2009 at 6:36 AM, Michael Collins m...@freeswitch.orgwrote:
On Wed, Nov 18, 2009 at 8:32 AM, mayamatakeshi
mayamatake...@gmail.comwrote:
About mod_fifo, it would be safe to use it in multi
On Tue, Nov 17, 2009 at 10:04 AM, t...@a2unlimited.com wrote:
MC,
Yes, I tried make make perlmod, which did not fix the error.
Just finished deploying an instance of the application on another server
that did not produce the error (exact same configuration).
Not sure what is causing it,
On Tue, Nov 17, 2009 at 8:22 AM, Jerry Richards
jerry.richa...@teotech.comwrote:
MC,
We would like the dialplan to route the call based on Presence, which is a
database lookup. I should be able to do this in Lua, true?
Jerry
Yes, you can use Lua for this if you wish to do so, HOWEVER,
Try doing this:
http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install
-MC
On Tue, Nov 17, 2009 at 7:12 AM, Sam Abekah-Mensah ab...@greatiam.comwrote:
Hello
I have tried the same setup but this time using a windows build FS1.0.4
on an XP machine and all is fine. The sample 1001 and 1002
On Sat, Nov 14, 2009 at 5:18 PM, Samuel Abekah-Mensah ab...@greatiam.comwrote:
Hello
Please pardon me if the solution to this is somewhere already that I
have been unable to locate. I have just got a straight out-of-the-box
build of FS. According to the wiki, I should be able to test using
FYI,
I've added the skeleton of the agenda for this week's call:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_11_20
The agendas have been pretty light lately. I would like everyone to think
about questions that could be brought up for discussion. Also, I'd like to
take this time to say thank
On Mon, Nov 16, 2009 at 9:36 AM, Jerry Richards
jerry.richa...@teotech.comwrote:
I have a bit of confusion about Lua scripting. When a script is invoked,
should it always return an XML string that is used by FS? Or as in the
case
of dialplan examples, does it actually execute the dialplan
We are still working on 1.0.5. Right now the best place to be is that latest
trunk. More information is forthcoming...
-MC
On Mon, Nov 16, 2009 at 3:07 PM, Tim Uckun timuc...@gmail.com wrote:
Where is 1.05? The trunk? Is trunk stable?
Thanks.
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