The password is set in conf/autoload_configs/event_socket.conf.xml
-MC
Sent from my iPhone
On Dec 20, 2009, at 7:58 AM, Joseph L. Casale jcas...@activenetwerx.com
wrote:
Trying to setup a new config in the pfSense 1.2.3 final package and
when
I try to connect to the console I get an
On Dec 17, 2009, at 11:34 PM, Jason White ja...@jasonjgw.net wrote:
Edmar Cruz darklio...@yahoo.com wrote:
Is there a link or tutorial for the expressions format.
Anything that describes Perl regular expressions should help, and for
reference, see the pcre(3) manual page, and use the
On Dec 12, 2009, at 7:28 PM, David V. Fansler dfans...@dv-
fansler.com wrote:
I am new to FreeSWITCH (ok a month old) and am still learning as
hard as I
can. In the recent talk about documentation, I had noticed that
finding
documentation on the FreeSWITCH wiki was a bit of a chore.
On Dec 12, 2009, at 7:41 PM, Robert Clayton rjca...@gmail.com wrote:
David,
Thanks for your hard work.
Maybe more organization will make the areas needing substance or
explanation more obvious.
That's my hope as well.
-MC
Bob
On Sat, Dec 12, 2009 at 10:28 PM, David V. Fansler
On Dec 11, 2009, at 4:02 PM, bcxml bc...@hotmail.com wrote:
The version is
FreeSWITCH Version 1.0.4 (14460)
Ouch. You are nearly 6 months and 1500 revs behind. You badly need to
update to latest trunk.
-MC
___
FreeSWITCH-users mailing
Version 1.0.5 pre 8 is due out any minute. Definitely upgrade to trunk
or at least pre8 when it's available.
-MC
Sent from my iPhone
On Dec 7, 2009, at 6:29 PM, DJB djbin...@yahoo.com wrote:
We have FreeSWITCH Version 1.0.4 (exported) running at a high volume
traffic. I normally check the
On Nov 6, 2009, at 3:59 PM, Dave Stevenson
steve...@primrosebank.net wrote:
Hi,
can someone pointme to where the valid dialing strings are specified ?
For SIP dialstrings check here:
http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings
Also, if you send us examples of
On Nov 6, 2009, at 3:21 PM, Orien Love or...@tx.rr.com wrote:
First of all, Thanks to the help I received on my pfSense
installation,
especially to Michael. I have a basic test system up and running. I
am
still waiting on some hardware but the base system is working
I am looking
On Nov 6, 2009, at 3:21 PM, Orien Love or...@tx.rr.com wrote:
First of all, Thanks to the help I received on my pfSense
installation,
especially to Michael. I have a basic test system up and running. I
am
still waiting on some hardware but the base system is working
I am looking
On Nov 6, 2009, at 3:21 PM, Orien Love or...@tx.rr.com wrote:
First of all, Thanks to the help I received on my pfSense
installation,
especially to Michael. I have a basic test system up and running. I
am
still waiting on some hardware but the base system is working
I am looking
I can guarantee that the FS devs are well aware of pj-sip. If it was/
is a viable alternative then it would be considered. The fact that it
isn't being used is a pretty good indication that it isn't suitable
for FS at this time.
-MV
Sent from my iPhone
On Oct 31, 2009, at 1:21 PM, Meftah
Judging by this error I would assume that you're still calling
sched_api as a Dialplan application and not as an FS API command.
You need to figure out how to create an API obj in java and call
sched_api from that object.
-MC
Sent from my iPhone
On Oct 21, 2009, at 2:44 AM, Henry Huang
Everyone repeat after me:
make current is my friend
make current is my friend
make current is my friend
ALWAYS use make current unless you know more about FreeSWITCH than the
project's authors do.
-MC
On Oct 14, 2009, at 8:56 PM, Brian West br...@freeswitch.org wrote:
Make current will NOT
On Oct 13, 2009, at 8:30 PM, Henry Huang red.rain.se...@gmail.com
wrote:
Diego:
You probably miss understood me. I said I was able to make
sched_hangup work, but not the sched_api in the same way I
script for sched_hangup
The problem was on the second paragraph.
Henry, can you
Gang,
Does anyone have a working analog setup? I've reproduced symptoms on
two different systems, one with the Rhino 8 port card and one with the
Nxtvox TDM400 clone. Symptoms are identical when dialing out on the
FXO port. The outbound dialing established a connection with the
called
have all been pretty successful. But thanks
for info! Anyone else using Rhino, Digium, or compatible analog cards?
Thanks!
-MC
On Sat, Sep 12, 2009 at 21:06, Michael S Collins
m...@freeswitch.org wrote:
Gang,
Does anyone have a working analog setup? I've reproduced symptoms on
two
I get the feeling that you are trying to use the wrong tool for the
job. If you need to launch a script after the call ends AND you need
access to the CSV file then you either should switch to XML CDR or
just write a Perl script that runs as a daemon that sits and waits for
CSV files to
On Sep 5, 2009, at 8:25 PM, Adam Wilt wiltingt...@gmail.com wrote:
Hi, the documentation says that mod_commands is available from
within mod_lua. But when I try to access it like this:
session:execute(uuid_broadcast,session_id .. .. filename ..
both)
I get: Invalid Application
Sent from my iPhone
On Sep 4, 2009, at 10:03 PM, Ujjval Karihaloo
ujj...@simplesignal.com wrote:
Would that be firewall on the CentOS machine that FS is installed on?
Correct
From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org
]
Sent from my iPhone
On Sep 3, 2009, at 6:08 AM, NOx-WHV enno.egb...@googlemail.com wrote:
Hi,
does anybody have a tip how to start a batchfile after hanging up.
After ext. 1000 calls 1001 and hang up, i need a request to call:
/../../FS/batchfile 1000
if 1001 calls 1000 i need:
Tom,
You should not have to do anything other than make samples. If you are
having these kinds of problems then something went very wrong during
the install. Try starting from scratch just to be sure. Also, what
distro are you running?
-MC
Sent from my iPhone
On Aug 29, 2009, at 8:40
Sent from my iPhone
On Aug 27, 2009, at 10:01 PM, lakshmanan ganapathy
lakindi...@gmail.com wrote:
No. In the dial plan I said, application=perl data=The perl
script.
I also checked $session-execute(bridge,user/1010). This is
working fine.
But originate is not working as I expected.
I have looked at that but I am confused on which files need to be
edited. Since I have already installed in Wanpipe mode with the
Sangoma card I skipped straight to the Wanpipe section. It mentions
setting the [span wanpie PRI_1] etc in the openzap.conf then further
down it mentions
Sent from my iPhone
On Aug 26, 2009, at 6:00 PM, Lars Zeb larc...@yahoo.com wrote:
Is there a way to dial an external 10-digit phone number, wait a
second or two after connecting, and then dial a 4-digit extension?
From the CLI:
originate sofia/gateway/mygw/1234567890 1234
As for the
Be sure to visit wiki.freeswitch.org and search for rosetta stone
which will take you to a page that helps translate Asterisk concepts
into FreeSWITCH concepts.
It will seem strange at first but once you get over the hump you will
really appreciate the power of FreeSWITCH. Be sure to visit
Jim,
Just curious - could you document this use case on the wiki? Maybe you
could create a page describing the setup and then link to it from the
TLS page.
Thanks,
MC
Sent from my iPhone
On Aug 5, 2009, at 9:07 AM, Jim Burke j...@evolutiontel.net wrote:
Hi NOx,
Can you clarify the
How about this: strftime
Try it at the CLI
-MC
Sent from my iPhone
On Jul 25, 2009, at 7:20 AM, Milena testeado...@gmail.com wrote:
Hello everyone,
I'm using the inbound event socket to receive some information about
the status of my FreeSWITCH system and i wanted to know if there is
What svn rev are you on? There have been some important changes
recently.
-MC
Sent from my iPhone
On Jul 19, 2009, at 5:11 AM, Mark Campbell-Smith mcampbellsm...@gmail.com
wrote:
Hi All,
I know this question has come up before but I couldn't find the answer
that I could understand!
On Jul 15, 2009, at 5:40 PM, freeswitch-users@lists.freeswitch.org
wrote:
Could you please just tell me where to set it??
The menu actions are defined in conf/autoload_configs/ivr.conf.xml
The audio played for the menus is defined in conf/lang/en/demo/demo-
ivr.xml
-MC
Please try
I wonder if it would make sense to create a separate sub-command like
show channels stats or something. That way we could put all sorts of
nifty info there without breaking the existing command.
Thoughts?
-MC
Sent from my iPhone
On Jul 16, 2009, at 7:37 AM, Apostolos Pantsiopoulos
On Jul 14, 2009, at 5:02 AM, Eli Hayun eliha...@gmail.com wrote:
Hi
I am not using fixed xml files for the extension registration. I have
LUA script to return an XML string to FS.
Everything goes fine until I am trying to get the voice messages.
When am entering my id, FS (or voicemail
IIRC you need to supply the uuid because the socket doesn't make any
assumptions about the APIs you send.
-MC
Sent from my iPhone
On Jul 11, 2009, at 2:21 PM, Brian West br...@freeswitch.org wrote:
I think you do ...
/b
On Jul 11, 2009, at 4:08 PM, Nik Middleton wrote:
Excellent. Do I
A few questions for you if I may:
FreeSWITCH doesn't yet have a GUI -are you okay with XML config files?
Do you have TDM circuits for your outbound traffic or are you using a
SIP provider?
BTW, mod_vmd is used to detect an answering machine beep, but it does
not detect human vs. machine. For
Look more closely at the output. It looks like mod_libpri.so didn't
get installed properly. I think this is a bug in the ozmod_libpri
build. For now just locate that missing .so file in your oz build
environment and copy it to the freeswitch/mod directory and try again.
-MC
Sent from my
If you're already on trunk then just do make current
-MC
Sent from my iPhone
On Jun 30, 2009, at 2:04 AM, Saeed Ahmad saeedahmad1...@gmail.com
wrote:
Hi,
What is the best way to update to latest version if we are already
running an older stable version?
I am using SVN trunk, sources
On Jun 30, 2009, at 8:43 PM, Edmar Cruz darklio...@yahoo.com wrote:
I have 1 Fs and 1 Asterisk if G729 is available on Asterisk so i
shall load
to G729 for freeswitch that needs a license?
You need a license if you are transcoding to or from G729. If you are
just passing the media
Couldn't you just throw all the calls into a conference at this point?
-MC
Sent from my iPhone
On Jun 26, 2009, at 9:30 PM, Matthew Fong mattdf...@gmail.com wrote:
can you 3 way with uuid_bridge?
--matt
On Fri, Jun 26, 2009 at 9:08 PM, Brian West br...@freeswitch.org
wrote:
Not sure what
On Jun 23, 2009, at 7:04 AM, Max Bridgewater
max.bridgewa...@gmail.com wrote:
Hi Michael,
Using loopback solves my problem. Thanks a lot.
There is a strange thing i observed though. I need to paste my
extension in the default.xml file. Having them in the default
directory isn't
FYI, the devs report that they are at the restaurant! Last chance to
pitch in and feed the troops. :) hit the paypal button on the main
FreeSWITCH page:
http://www.freeswitch.org
Keep those devs happy and fed and version 1.0.4 will be here before
you know it!
-MC
Hehe, where can I buy stock in this company? :)
-MC
Sent from my iPhone
On Jun 15, 2009, at 4:33 AM, David Knell d...@3c.co.uk wrote:
On Wed, 2009-06-10 at 10:02 -0700, Paul Mahler wrote:
What is the current status of Freeswitch? Can I safely use it in a
large scale commercial environment?
Just curious, why are you dialing out the external gw?
-MC
Sent from my iPhone
On Jun 13, 2009, at 11:11 PM, Edmar Cruz darklio...@yahoo.com wrote:
Ip Windows: 192.168.0.104
Ip Linux:192.168.0.105
My windows:
My sample on sip_profiles/external/dialus.xml
extension name=dialus
I will clarify the point on the wiki since it is a bit inaccurate
-MC
Sent from my iPhone
On Jun 10, 2009, at 10:40 AM, Brian West br...@freeswitch.org wrote:
Yes.
/b
On Jun 10, 2009, at 12:34 PM, Ben Jones wrote:
Does this mean if a user is set for rfc2833 OR info that FS will
generate
You have a few choices. If you can hold out until g729 licensing is
available in FreeSWITCH then that's a viable option. If not then
you'll need to go the hardware route. Go to wiki.freeswitch.org and
search for mid_dahdi_codec and you can learn more about the details.
Essentially you can
Did you try clean solution as brian suggested?
-MC
Sent from my iPhone
On May 23, 2009, at 8:33 AM, mszla...@aol.com wrote:
I get a lot of these errors on vc++ express 2008:
59Linking...
59LINK : fatal error LNK1181: cannot open input file '..\..\..\..
We are pretty much booked solid as we've got some unconfirmed speakers
we haven't posted yet. I'm redoing the schedule and will have an
updated one out this next week. One thing that we really need is
backup speakers. Our experience is that there are always people who
have emergencies and
Do you mean from the CDR? I recommend XML CDRs because they give tons
of information. If you are talking about gathering this stuff midcall
then you'll need to supply more information about your setup.
-MC
Sent from my iPhone
On May 10, 2009, at 6:17 PM, Diego Toro dft...@yahoo.com wrote:
If you installed the default configuration then you've already got a
PBX. I recommend looking at the default.xml file in conf/dialplan and
also the user setup files like 1000.xml in conf/directory/default.
The default config has 20 users already setup including voicemail. It
also has
On Apr 1, 2009, at 6:59 AM, Raymond Chandler
intralan...@freeswitch.org wrote:
seven wrote:
I know that. And I'd like to read code. Developers written great code
and also plenty of comments(which is documentation) in code. However,
there are sth. don't need to comment in code but should be
On Mar 31, 2009, at 5:59 AM, Anthony Minessale anthony.miness...@gmail.com
wrote:
if you set the channel variable 'session_in_hangup_hook=true' early
in the call, the session will be present in your script.
Very cool. I will get this chan var documented on the wiki right away.
-MC
On Mar 30, 2009, at 1:34 AM, François Delawarde fdelawa...@wirelessmundi.co
m wrote:
Thanks for your quickclear answers.
On Fri, 2009-03-27 at 10:55 -0500, Anthony Minessale wrote:
If you are using on-hook agents, it will place as many outbound calls
as there are people waiting.
If you
On Mar 11, 2009, at 8:34 PM, Brian West br...@freeswitch.org wrote:
;) I expect to see you at cluecon this year?
/b
Notice how he threw in a compliment and an invite to CC but didn't
actually address the question? ;) pretty sneaky bkw!
BTW, the best way to come to CC is to get your boss
On Feb 19, 2009, at 7:53 PM, Jason White ja...@jasonjgw.net wrote:
I have it working now. The relevant changes were as follows.
action application=set data=hangup_after_bridge=true/
action application=set data=continue_on_fail=true/
action application=bridge
Sent from my iPhone
On Feb 7, 2009, at 9:03 PM, Paul D. pa...@versafon.com wrote:
Followed Wiki to install and configure mod_cepstral. The problem is FS
always defaults to one voice, which I installed first, and ignores
others.
I did define SWIFT_HOME and added swift lib path to
Print out the variable to make sure it is what you expect:
io.write(argv is .. argv[1] .. \n;
Also, if you don't give the sound file an absolute path name then it
will automatically use the sound dir path.
-MC
Sent from my iPhone
On Feb 8, 2009, at 2:31 PM, Nik Middleton
Nik,
I see your point about the wiki entry regarding luasql. If someone
could clarify then I will be happy to help get the wiki documentation
updated appropriately.
-MC
Sent from my iPhone
On Feb 8, 2009, at 2:41 PM, Nik Middleton nik.middle...@noblesolutions.co.uk
wrote:
Hi Guys
On Feb 6, 2009, at 6:27 PM, Michael Jerris m...@jerris.com wrote:
Not sure what you mean by playback speed. All the prompts for
voicemail are defined in the phrase macros in the configuration.
Check out conf/lang/en/vm/sounds.xml
-MC
___
Out of curiosity which SIP messages have you been watching for on the
event socket? Also, how are you connected to the event socket? Are you
subscribing to all events and sifting through them to confirm that no
events are being fired when SIP messages are being sent?
-MC
Sent from my iPhone
On Jan 22, 2009, at 1:11 PM, Gregory Boehnlein da...@nacs.net wrote:
You use it on your own risk
Also, G.729 is patent encumbered big-time. Instead of lining the
pockets of lawyers and mega-corporations by perpetuating the use of a
crusty old codec we should all twist arms and get our
On Jan 22, 2009, at 1:55 PM, Kristian Kielhofner kristian.kielhof...@gmail.com
wrote:
On Thu, Jan 22, 2009 at 4:40 PM, Michael Collins
m...@freeswitch.org wrote:
On Thu, Jan 22, 2009 at 1:35 PM, Brian West br...@freeswitch.org
wrote:
Speex, while nice I think it would use more
Can you join irc later today? I will be on as mercutioviz. I would
like to discuss this more.
-MC
Sent from my iPhone
On Jan 20, 2009, at 2:10 AM, Krzysztof Zimnicki krzys...@go2.pl
wrote:
Can you post your openzap.conf file?
-MC
Sure.
[span zt]
name = OpenZAP
number = 1
Tamas,
The channel variable won't work for you if you can't ignore early
media. Your best bet is to use the variable execute_on_answer to
transfer an answered call to a new extension. Then you could just
sleep for 15sec and then check the value of endpoint_disposition.
What is the
On Jan 15, 2009, at 1:02 PM, Brian West br...@freeswitch.org wrote:
On that note the OpenVZ instances could live migrate from box to box
without dropping calls and usually had a small acceptable blip in
audio.
I'd say a small blip is quite acceptable compared to the alternative!
-MC
/b
Could you please do a backtrace and post it to a pastebin? If in Linux
do this:
gdb /path/to/freeswitch /path/to/corefile
-MC
Sent from my iPhone
On Jan 12, 2009, at 5:23 AM, shehzad p pmh...@gmail.com wrote:
Hi all,
I am also testing FS release 1.0.2, but I faced strange problem.
When
This looks like a zaptel issue. Do we have any zaptel users familiar
with this issue? If not you should probably check on the asterisk list
for help. Once zaptel is working we can then see about getting openzap
up.
-MC
Sent from my iPhone
On Jan 12, 2009, at 6:36 AM, Franziska Röhler
mode on in freeswitch?
Can it be done from cli on fly?
If I put debug mode on, will it affect performance of the
freeswitch, as the
freeswitch is currently used as production system.
Michael S Collins wrote:
So the issue is not happening right now? If it is then we would want
you
Please try latest first. If you still have errors then put them on
pastebin.
-MC
Sent from my iPhone
On Jan 9, 2009, at 11:36 AM, Tim B timb0...@hotmail.com wrote:
MC was using 1.0.2. I don't have access to the machine right now.
I can post the errors later tonight or try to build with
:)
Sent from my iPhone
On Jan 9, 2009, at 7:38 AM, Peter P GMX prometheus...@gmx.net wrote:
Hello Michael,
sorry for the inconvenience. It turned out that our Telco had to reset
the second PRI line. Now it works.
Best regards
Peter
Peter P GMX schrieb:
Hello Michael,
here is a log
Dude, you ROCK!
-MC
Sent from my iPhone
On Jan 9, 2009, at 3:06 PM, Andrew Thompson and...@hijacked.us wrote:
Hi,
Today I'd like to announce the open-sourcing of a distributed
callcenter platform I've been designing/building using
FreeSWITCH/Erlang. The goal is to allow multiple callcenter
Jeng,
Your condition expressions are not right. Could you describe what you
hope to accomplish with those two expressions? Once you get the
regular expressions figured out then it should all work.
-MC
Sent from my iPhone
On Jan 8, 2009, at 6:44 PM, Jian Yuan Peng jyp...@yahoo.com wrote:
What FS revision and is this a default Dialplan? Please pastebin the
output of the CLI while making test calls. Be sure to press F8 to
enable debug messages. Also, if you can do so turn on SIP messages by
launching FreeSWITCH with TPORT_LOG=1.
-MC
Sent from my iPhone
On Jan 6, 2009, at
A jira for the core dumps would be good, especially if you can
reproduce the behavior.
Question: where does mod_vmd come into play?
-MC
Sent from my iPhone
On Dec 29, 2008, at 6:28 AM, Adam Wilt wiltingt...@gmail.com wrote:
Should I add this to Jira?
On Sat, Dec 27, 2008 at 9:15 PM,
For the sake of testing can you record a call that gets a busy signal?
At least then we could analyze the audio and see what's going on.
If you need a dialplan example for this let me know.
-MC
Sent from my iPhone
On Dec 29, 2008, at 5:04 AM, Baskar yudha2...@gmail.com wrote:
Hi Michael,
I wonder if putting a sleep statement in your shell script might help.
If it's a timing issue then possibly the shell script is trying to
access the file before FS and/or the OS are done with it. You would
need to tinker with how long to sleep in order to find a value that
works in all
Phil,
Can you do the same test with debug turned on? F8 or console loglevel
debug will do the trick.
-MC
Sent from my iPhone
On Dec 25, 2008, at 12:38 PM, can_...@gmx.de wrote:
Hello,
I am trying to replace some static settings with dynamic ones which
are provided by a webserver. I can
Carole,
Are you calling the hangup app from the Dialplan?
-MC
Sent from my iPhone
On Dec 23, 2008, at 7:04 AM, Brian West br...@freeswitch.org wrote:
Well in this context the phones need to hangup... they aren't going to
do so automatically. So you'll need to hang up on them or they will
Have you checked out 'sched_api'?
-MC
Sent from my iPhone
On Dec 23, 2008, at 4:25 AM, Alexandru Nedelcu a...@sinapticode.ro
wrote:
Hi,
When I make a unsuccesfull call using session.originate, I'd like to
have a 10 minutes pause and then try again.
For our dialer we are using JS
I'm pretty sure that this is doable. Could you give us a hint as to
what arguments you want to send? For example, do you have one or more
channel variables you'd like to pass to the shell script?
-MC
Sent from my iPhone
On Dec 23, 2008, at 6:25 PM, Jason White ja...@jasonjgw.net wrote:
I am not a lawyer so I can't tell you for sure. However, I'm not aware
of any US laws against beep detection.
-MC
Sent from my iPhone
On Dec 21, 2008, at 7:09 AM, Gopalakrishnan A.N sai...@gmail.com
wrote:
Hi Micheal,
Is it anything like i am violating the laws? please let me
You've got ignore_early_media set to true but busy signals might be
sent during early media. Why are you ignoring early media?
Also, you might need to check your tone_detect syntax. You're set to
detect 400Hz but you haven't told the system what to do if it does
detect that tone. Please
Check out this page:
wiki.freeswitch.org/wiki/Debugging_Freeswitch
-MC
Sent from my iPhone
On Dec 18, 2008, at 6:38 AM, pe...@networkoblivion.com
pe...@networkoblivion.com
wrote:
What is the process for capturing and submitting a core dump?
I am messing around with the Cisco 79x1 phones
= (switch_memory_pool_t *) 0x80528f0
So I would guess it is trying to access an invalid memory location,
but
why, I have no idea
Any ideas?
Peder
Michael S Collins wrote:
Check out this page:
wiki.freeswitch.org/wiki/Debugging_Freeswitch
-MC
Sent from my iPhone
On Dec 18, 2008
Helmut,
Can you turn on full debug and capture the output? It's a lot so put
it in a pastebin.
-MC
Sent from my iPhone
On Dec 17, 2008, at 7:30 AM, Helmut Kuper helmut.ku...@ewetel.de
wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Anthony,
thx, but that doesn't work very
By ignore do you mean filter out? Or do you mean don't do anything but
do audiblize the tones? Do you have some sort of application that does
something with dtmfs?
-MC
Sent from my iPhone
On Dec 17, 2008, at 10:00 AM, stephen at stephenjc step...@stephenjc.com
wrote:
I have a click to
Just a hunch but try removing the spaces in this line:
loadzone=de
Zaptel can be quirky.
-MC
Sent from my iPhone
On Dec 16, 2008, at 5:54 AM, fidibus83 fidibu...@aol.com wrote:
I have installed zaptel-1.4.11
I have looked in zonedata.c and there is configured de-tonezone
Von:
to have to get this mesage
decoded for debugging reasons.
regards
helmut
Am 27.11.2008 19:02, schrieb Michael S Collins:
You can ignore this one for now. Eventually this will be handled but
it shouldn't affect your calls. I've been ignoring it for six
months. :)
-MC
Sent from my
Do you need something just for one extension? Or system wide?
If it's system wide then all you need is an extension that matches a
condition like this:
condition field=destination_number expression=^1$
Anyone who dials just a single digit 1 will go to this Dialplan entry.
From there you
Ccav,
Thanks for helping out with the project! If you haven't already joined
us on the irc channel please do so: #freeswitch on irc.freenode.net.
Another channel you might be interested in is #tcapi. There is a group
working on a general purpose GUI for FreeSWITCH at tcapi.org. They've
Jason,
Thanks for pointing this out. You are correct. This is a case of
development moving faster than documentation efforts. I will update
the wiki.
-MC
Sent from my iPhone
On Dec 13, 2008, at 10:23 PM, Jason White ja...@jasonjgw.net wrote:
The Wiki page at
Faisal,
Dennis makes a good point: you are mixing event socket syntax with
dialplan syntax. I recommend starting with the dialplan example on the
wiki. To get a single sound file to play over and over put it in a
directory by itself.
Also, there is an undocumented feature called .loc files
Check out mod_localstream on the wiki and see if that sounds like what
you need. I'm still learning it all myself but I believe that's where
you should start. Please report back with any questions and we will
take it from there!
-MC
On Dec 5, 2008, at 3:48 AM, Faisal Maqsoodi [EMAIL
What would need to happen after the tone is sent back out? Also, would
this be part of something like an IVR?
-MC
On Dec 5, 2008, at 7:22 AM, Frank @ Impact [EMAIL PROTECTED]
wrote:
Is there any dialplan instructions that could be added that would sit
and listen during a call for a tone
I will do some research on this and let you know what I find out.
Question: are these internal calls or pstn or ?? Just curious about
your environment.
Thanks,
MC
On Dec 5, 2008, at 4:23 AM, Apostolos Pantsiopoulos [EMAIL PROTECTED]
wrote:
The proto_specific_hangup_cause is missing on
What is your originate string?
-MC
On Dec 5, 2008, at 3:54 AM, Peter P GMX [EMAIL PROTECTED] wrote:
I am building an IVR application where an incoming call is
initiating an
outgoing call. When I pass a variable_other_uuid (the uuid of the
incoming channel) at originate time, I am able to
Evgeniy,
I will need some time to digest all of this. I have an a104 but I
don't have a solaris system for testing. I will report back as soon as
I can.
-MC
On Dec 5, 2008, at 6:53 AM, Evgeniy Zolotov [EMAIL PROTECTED] wrote:
Greetings!
Question about possibility of the use
Will the call be terminated at that point or does it need to continue?
I do know that the tone_detect app can listen for a dtmf from either
direction and can trigger execution of another app/extension/etc.
However, I've never tried it on a bridged call, so I'm curious to see
what would
specific hangup cause? Otherwise it would be difficult to understand
what really happened.
Michael S Collins wrote:
I will do some research on this and let you know what I find out.
Question: are these internal calls or pstn or ?? Just curious about
your environment.
Thanks,
MC
On Dec 5
That's a pretty old rev. Any chance you could make current?
-MC
Sent from my iPhone
On Dec 5, 2008, at 5:09 PM, Frank @ Impact [EMAIL PROTECTED]
wrote:
I tried your suggested test. Here is the business end of the
extension
I tried.
action application=set data=DTMF1=false/
Gopal,
FreeSWITCH does not have free amd but you can buy a license. Please
send a request to [EMAIL PROTECTED] I have some experience
with running amd so I can assist with setup questions.
-Michael
Sent from my iPhone
On Dec 4, 2008, at 4:50 AM, Gopalakrishnan A.N [EMAIL PROTECTED]
Does the core dump always happen in this call scenario? If so, can you
get a back trace? Put it on pastebin. That will hopefully help narrow
down the issue.
-MC
Sent from my iPhone
On Dec 1, 2008, at 11:27 PM, Baskar [EMAIL PROTECTED] wrote:
Hi,
I have updated all the above events you
You can ignore this one for now. Eventually this will be handled but
it shouldn't affect your calls. I've been ignoring it for six months. :)
-MC
Sent from my iPhone
On Nov 27, 2008, at 9:27 AM, Peter P GMX [EMAIL PROTECTED] wrote:
I have installed OpenZAP with a TE220 card and EuroISDN.
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