Many audio formats let you embed meta data with this kind of
information. That's how you can hover your mouse over a wav file in
Windows and it pops up a little box that says, "Artist: Britney Spears,
Title: Hit Me Baby One More Time, etc."
-MC
> -Original Message-
> From: [EMAIL PROTECTE
Guys,
I know how much you all just *love* SIP, so I thought you might be
interested in chiming in on this article:
http://blogs.zdnet.com/carroll/?p=1877
-MC
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Freeswitch-users@lists.freeswitch.org
http://lists.fr
Jon,
Are you saying that you cannot pass a comma-separated list of variables
like this?
Note: no spaces!
-MC
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jon
Bruel
Sent: Tuesday, September 23, 2008 12:32 PM
To: freeswitch-users@lists.freeswi
I use single quotes like this:
data="[var1='Michael Collins',var2='FreeSWITCH enthusiast',var3='you get
the idea']"
-MC
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Wasim Baig
Sent: Tuesday, Se
Hmm... let me lab up this scenario and see if I can't figure out what's
up.
-MC
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jon
Bruel
Sent: Tuesday, September 23, 2008 2:55 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswit
Jair,
Are you meaning to call an extension, and if the called party doesn't
answer, go to his/her voicemail? If so you probably want to do a bridge
app with a timeout value. Check out:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall#Timeout
So you set a timeout, then bridge
Transferring to "1000" will send the call to another extension
altogether - it won't be in "public_did" but rather whatever extension
matches "destination_number" with a value of "1000" - no?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jair
S
linked to the extension, and this DID is
called, the VM doesn't work.
Jair
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Michael Collins
Sent: Wednesday, September 24, 2008 4:17 PM
To: freeswitch-users@lists
lf Of
Michael Collins
Sent: Wednesday, September 24, 2008 4:38 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Voicemail x DID
Does x1000 ring when you do the transfer app in your
"public_did" extension? Possibly you need to tra
> I'm ready to tackle creating a multi-level IVR. From what I have seen
in
> the
> wiki, what I have to do is to create a separate wav file for each
prompt
> in
> some other software and then create an XML file to organize the logic
> around
> pointers to the wav files and terminating in the requi
Very interesting! I noticed that there are three similarly-named apps on the
wiki that could use some love:
set_global
set_profile_var
set_user
Could someone in the know throw some light on these three? I'll be happy to
wikify anything that gets posted to the list.
-MC
On Thu, Sep 25, 2008 at 12:
Anish,
There is a book in the works but it is very, very preliminary. Earliest
possible publication wouldn't be until late 2009...
-MC
On Thu, Sep 25, 2008 at 1:37 PM, Anish Basu <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I noticed that the documentation efforts have been ramped up and that the
> wik
To add to Brian's thought...
The "answer event" that you receive does have lots of information. You
are interested in only one piece of that information, no? You'll need to
parse the reply and grab only the line that contains "Answer-State:
x"
Did you say that you are using PHP for this
You need to handle each response from the server, no? Can you post your
PHP code here?
In Perl I would do something like this.
# $data contains CHANNEL_ANSWER event stuff
if ( $data =~ m/Answered-State: (\w+)/m ) {
my $state = $1;
print "Channel state is $state\n";
if ( $state eq 'answ
er 03, 2008 11:07 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] get channel status
Please find the attached PHP file
On Fri, Oct 3, 2008 at 11:29 PM, Michael Collins
<[EMAIL PROTECTED]> wrote:
You need to handle each response from the server, no? Can you po
Regex 101: use parentheses to capture info in $1:
^(234)$
$1 will = '231'
-MC
> -Original Message-
> From: [EMAIL PROTECTED]
[mailto:freeswitch-
> [EMAIL PROTECTED] On Behalf Of Peter P GMX
> Sent: Friday, October 03, 2008 4:14 PM
> To: freeswitch-users@lists.freeswitch.org
> Subject: [F
Based upon your original post I'd say you probably want mod_xml_curl,
which essentially just fetches data from a server. I believe you will
find some examples in the source directory under
scripts/contrib/intralanman/PHP/fs_curl
-MC
From: [EMAIL PROTECTED]
In your sip profile add the params
Just a quick question - is that a typo? Shouldn't it be "rfc1918" and
not "rfc918"? I want to confirm this so that we can get it properly
documented on the wiki.
-MC
___
Freeswitch-users mailing list
Freeswitch-
> I'm building a web interface with Python/Django.
>
> Freeswitch will run on a separate server and fetches the information
> using
> xml_curl. That's working fine.
>
> What I want to do is:
>
> I want that for every voicemail received, freeswitch uploads it to
another
> server, using some kind
Check the src directory: docs/phrase/phrase_en.xml
> -Original Message-
> From: [EMAIL PROTECTED]
[mailto:freeswitch-
> [EMAIL PROTECTED] On Behalf Of Alex Kinch
> Sent: Thursday, October 09, 2008 2:23 PM
> To: freeswitch-users@lists.freeswitch.org
> Subject: [Freeswitch-users] Sound file
Dude, let me answer the easy ones every once in a while!! :P
> -Original Message-
> From: [EMAIL PROTECTED]
[mailto:freeswitch-
> [EMAIL PROTECTED] On Behalf Of Brian West
> Sent: Thursday, October 09, 2008 2:28 PM
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-user
Thanks for putting comments in the code. Is there possibly a place on
the wiki where this script might be appropriate?
-MC
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah
Silverman
Sent: Thursday, October 09, 2008 3:11 PM
To: Freeswitch-use
> I've started working on a wiki for SIP documentation for interop and
> other features. I've created a basic page for Freeswitch:
Kristian,
We know you are an active member of the Asterisk community so we thank
you for showing FS a little love! We appreciate it when OSS telephony
users go the
> Can I at least be a beta tester or something? Please? I'm
desperate!!!
Dude, you're hired! :)
-MC
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http:
> Yes, using the event socket;
>
> http://wiki.freeswitch.org/wiki/Mod_event_socket
>
Just another little thought: with the event socket you can connect to FS
and send pretty much any FS API command you want. You also can subscribe
to events and receive all sorts of cool information about what's
> Tested, and works. Once again Mike - MONEY!
Dontcha just *love* it when the devs respond? :) All they ask for is an
occasional jira to be filed and that you test it. Of course, they also
love it when you wikify anything that you have them add for you. (HINT
HINT)
Thanks for playing with FS - y
It's all in mod_xml_rpc.
I've been trying to get it all doc'd but I'm way behind... :-) Of
course, I'm the guilty party for putting "webapi" in the mod_perl page
on the wiki. Sadly, "webapi" shows up only on this page which means I'm
slacking. This page needs some love:
http://wiki.freeswi
> I'm far away from the default config, so I just want to add the
> MINIMUM amount necessary to my config to enable speech.
>
> My goal is to have some basic functions like "You have 3 dollars and
> 27 cents left".
Here's a snippet that I would use to manually create the above phrase,
mixing TTS
> However I believe that
> there are more elegant ways of handling dollar amounts when using the
> "say" action. Brian, can you confirm if "say" handles currency and if
it
> handles dollars/cents with correct plural/singular values?
Check out http://wiki.freeswitch.org/wiki/Speech_Phrase_Managemen
In other words, you don't have your extension in the right spot.
Check to see that it isn't in the default.xml dialplan file after the
enum extension. It needs to be *before* the enum extension because the
enum extension is the catch-all - it grabs everything that hasn't
already been matched i
: Re: [Freeswitch-users] How to get DISA working ?
This hint only works in the default configs in SVN trunk as of the past
two weeks if I recall. Also in your version you'll be
conf/diaplan/extensions/
/b
On Oct 16, 2008, at 2:30 PM, Michael Collins wrote:
HINT: You might wa
Very interesting. Could you point out those params and when they apply?
Also, does FS differentiate between an attended and a blind transfer?
Just curious if the type of transfer matters or not.
-MC
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behal
Scratch that last... I clicked send before thinking... :P
-MC
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Michael Collins
Sent: Thursday, October 16, 2008 4:12 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] How
> Open up the sip profile and make sure you apply the correct ACL and
> it'll work.
>
> /b
Brian, is this the relevant wiki entry?
http://wiki.freeswitch.org/wiki/Acl#sip_profiles
Just confirming.
Thanks,
MC
___
Freeswitch-users mailing list
Freeswi
Could you post a sample from an Asterisk dialplan and maybe we can come up
with a good way to replicate it in FS? Possibly there is a more elegant
solution that does not require lots of munging.
-MC
On Mon, Oct 20, 2008 at 12:48 PM, Kristian Kielhofner <
[EMAIL PROTECTED]> wrote:
> In Asterisk on
> Sure:
>
> [include]
> exten => 727,1,DoStuff
> exten => 800,1,DoOtherStuff
>
>
> [default]
> include => include
>
> I know it's a simple case and not really an "example" but I'm really
> just looking for a way for a context to just "plow through" another
> context looking for a match before
> > Your best bet is to try to shift the paragidm in your head away from
how
> > asterisk does it =D
> >
>
> I know... ATM the moment I'm trying to bring FS into the organization
> in stages, "parallel" to Asterisk as much as possible. Once I get
> through this initial phase I can begin to thin
Do you have wanpipe.conf and zt.conf in your conf directory? Are there any
permissions issues that need to be addressed?
-MC
On Tue, Oct 21, 2008 at 8:25 AM, Sias Mey <[EMAIL PROTECTED]> wrote:
> It would seem that way... since it is openzap complaining when it starts
> up.
>
> and there is a mod
So you need to create a second call leg that is somewhat independent of the
first leg, so that you can play a file, and *then* bridge the new leg to the
"current" leg?
I just want to make sure that I grok what you are trying to accomplish.
Also, if you haven't put your dialplan and script in paste
rd.
>
> Thanks,
> Sias
>
> On Tue, Oct 21, 2008 at 08:49:47AM -0700, Michael Collins wrote:
> >Do you have wanpipe.conf and zt.conf in your conf directory? Are there
> >any permissions issues that need to be addressed?
> >-MC
> >
> >On Tue, O
On Tue, Oct 21, 2008 at 10:43 AM, Anthony Minessale <
[EMAIL PROTECTED]> wrote:
> api break will break one of them just not all. it has nothing to do
> with event lock
> event lock means do not parse events recursively like the behavior you
> described.
>
>
> Since you asked so nicely, I added th
On Wed, Oct 22, 2008 at 9:50 AM, Thomas Troesch <[EMAIL PROTECTED]>wrote:
> Is there a document or recommended strategy on how to migrate from an
> existing PBX system to FS? There are 2 T1-PRI lines used 24/7 (call
> center) which can't be down for development or testing. The existing
> system
> > please please join irc and let the whole group help you.
> I was there today, but it seemed, that nobody could hear/read me. I
> will try again tomorrow.
What is your IRC nick? Mine is mercutioviz. I'm interested in this issue
because I've been dialing in some somewhat similar scenarios and I
FYI, check this page out:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_execute_extension
Just remember the difference between "transfer" and "execute_extension"
Transfer will go and never come back
Execute_extension will go "run" an extension like a subroutine and then come
back
Happy dia
It depends on exactly what your scenario is. However I've had good
success using the "hangup_cause" channel variable. From some of my
XMLCDRs I have values like these:
UNALLOCATED_NUMBER
NORMAL_CLEARING
DESTINATION_OUT_OF_ORDER
Hope that helps...
-MC
_
> Hello,
>
> I receive the following message during CS_INIT
>
> *Failed to load library libceplang_de.so due to:
> /opt/swift/lib/libceplang_de.so: undefined symbol: cst_rx_int*
>
Hmm... that's odd. Did you have an older version of Cepstral installed
prior to going to 5.1?
-MC
___
| 5.1.0 | Yes | female | 35 | German | 16000 Hz
>
> I googled around but couldn' find any hints to the cst_rx_int error.
>
> Best regards
> Peter
>
> Michael Collins schrieb:
> >> Hello,
> >>
> >> I receive the following message during CS_INI
gt;
> libceplex_de.so.5.1
> lrwxrwxrwx 1 root root 19 2008-10-28 23:26 libceplex_en.so.5 ->
> libceplex_de.so.5.1
> lrwxrwxrwx 1 root root 15 2008-07-17 18:10 libswift.so ->
libswift.so.5.1
> lrwxrwxrwx 1 root root 15 2008-07-17 18:10 libswift.so.5 ->
> libswift.so.5.1
>
> Yes, I agree. But one could use the two methods combined (csv or xml +
> db) for redundancy.
>
> Is there any consideration regarding automatic log rotation (e.g.
> hourly, or user specified)
> without the need of a HUP? Now, that could make things a lot easier
for
> the development of
> an exte
ts disk (or whatever storage) if the db itself went
down but the webserver stayed up.
Just a thought, anyway. It may be extra layers but it's also extra
control.
-MC
> Michael Collins wrote:
> >> Yes, I agree. But one could use the two methods combined (csv or
xml +
> >
/me sends Anthony's post to the printer to be laminated and framed...
:-)
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Anthony Minessale
Sent: Thursday, October 30, 2008 6:10 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitc
On Mon, Nov 3, 2008 at 4:12 AM, Andrew Gilbert <[EMAIL PROTECTED]>wrote:
>
> I am just an observer of FS. What I am impressed with is the
> extensibility. And I am curious about xml models (vxml, proprietary xml) vs
> scripting (lua, python, etc). That is why I asked.
>
I'm sure they're all poss
FYI,
There is like a 1400 line limit in pastebin, or something like that, so
you have to be careful when you have a large dump to put there.
-MC
> -Original Message-
> From: [EMAIL PROTECTED]
[mailto:freeswitch-
> [EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
> Sent: Tuesday, November
Mitul,
I recommend talking to s3g_fault on the #openzap channel. He is very clever
and helped me get over the hump with a Sangoma A104D card. The FS community
definitely needs comprehensive documentation for getting all of this stuff
set up and I promise that I will work on it all but I won't have
Well, MikeJ and anthm added RBS signaling support some time back. I could
get RBS to work in that I got all green lights and I could see the A and B
bits flipping up and down when I called a channel, but OpenZAP didn't
actually have the meanings of those A and B bits programmed in the
signaling. I
do you need to do a load mod_rbs? Or is it part of the OZ package?
On Fri, Nov 14, 2008 at 10:36 AM, Michael Jerris <[EMAIL PROTECTED]> wrote:
> There is a different mod now for rbs as well in openzap.
>
> On Nov 14, 2008, at 1:27 PM, Deepak wrote:
>
> > Thanks Michael. I will go with Sangoma.
>
under & CEO,
> Enterux Solutions Pvt Ltd,
> The Enterprise Linux Company(r),
> http://www.enterux.com/
>
>
> On 14-Nov-08, at 23:31, "Michael Collins" <[EMAIL PROTECTED]> wrote:
>
___
Freeswitch-users mailin
figuration.
-MC
On Fri, Nov 14, 2008 at 11:39 AM, Deepak <[EMAIL PROTECTED]> wrote:
> I can't find any module mod_rbs or something similar in either freeswitch
> 1.0.1 or SVN repository for OZ. Where do I find it? Thanks
>
> On Fri, Nov 14, 2008 at 1:43 PM, Michael Collins &l
I believe this is a limitation in JavaScript. There is no "sleep" command or
function, just some setTimeout kind of thing where you launch a function
after a predetermined interval. I think with js you're stuck with an
inelegant solution... :(
-MC
On Fri, Nov 14, 2008 at 2:42 PM, Jon Bruel <[EMAI
Try eval
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eval
-MC
> -Original Message-
> From: [EMAIL PROTECTED]
[mailto:freeswitch-
> [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
> Sent: Wednesday, November 19, 2008 10:56 AM
> To: freeswitch-users@lists.freeswitch.org
> Subject
On Thu, Nov 20, 2008 at 1:50 AM, Birgit Arkesteijn
<[EMAIL PROTECTED]> wrote:
> Hi Michael,
>
> Sorry to nag you, but did you manage to experiment with the Javascript
> session in this capacity?
> I hope you reached your office. :-)
>
> Cheers, Birgit
>
>
Sorry, day job was rough yesterday and I di
On Thu, Nov 20, 2008 at 6:58 AM, Brian West <[EMAIL PROTECTED]> wrote:
> ${sofia_contact([EMAIL PROTECTED])} will return "error/user_not_registered"
>
> /b
>
Side note: if you wanted to check from the CLI then is this correct?
sofia status profile internal reg [EMAIL PROTECTED]
Or is there bette
HINT: even though it takes a few minutes longer, it is advisable to
use "make current" because it will make 100% sure that you are on the
latest SVN, no build skew, etc.
-MC
On Thu, Nov 20, 2008 at 7:18 AM, Anthony Minessale
<[EMAIL PROTECTED]> wrote:
> did you actually update again.
> I am almost
On Thu, Nov 20, 2008 at 7:22 AM, Brian West <[EMAIL PROTECTED]> wrote:
> "sofia_contact [EMAIL PROTECTED]" will also work at the CLI :P
>
> /b
I swear I tried that and got an unknown command! Of course, when I
tried it again it worked perfectly. My punishment will be to put this
in the wiki somew
On Thu, Nov 20, 2008 at 1:50 AM, Birgit Arkesteijn
<[EMAIL PROTECTED]> wrote:
> Hi Michael,
>
> Sorry to nag you, but did you manage to experiment with the Javascript
> session in this capacity?
> I hope you reached your office. :-)
>
> Cheers, Birgit
Birgit,
After reviewing my dialplan I realize
Can you use the tags to handle the non-matching case?
-MC
> -Original Message-
> From: [EMAIL PROTECTED]
[mailto:freeswitch-
> [EMAIL PROTECTED] On Behalf Of henkoegema
> Sent: Friday, November 21, 2008 8:15 AM
> To: freeswitch-users@lists.freeswitch.org
> Subject: [Freeswitch-users] Nest
On Tue, Nov 25, 2008 at 11:44 AM, Anthony Minessale <
[EMAIL PROTECTED]> wrote:
> iirc call_timeout has been changed to originate_timeout
>
That seems to jive with what's in switch_ivr_originate.c as there is no
mention of "call_timeout" anywhere in the function switch_ivr_originate(),
but "origi
>
>
>
> I used "call-timeout" because that's how I understood it from the Wiki.
> (?)
>
Yep, that's all that there is on the wiki. Unfortunately the channel
variables page is one of many in need of some attention. I will add
"originate_timeout" right away. The only question remaining is what, if
I have some time...
-MC
On Tue, Nov 25, 2008 at 1:32 PM, Michael Collins <[EMAIL PROTECTED]> wrote:
>
>>
>> I used "call-timeout" because that's how I understood it from the Wiki.
>> (?)
>>
>
> Yep, that's all that there is on the wiki.
One way to help narrow it down is to revert back to an earlier working
version. In fact, if you can find the tipping point where it went from
working to not-working then that would really help narrow things down.
Do you know the last rev that actually worked? Looking at the change
logs I see a fai
http://wiki.freeswitch.org/wiki/Main_Page#Bounties
On Wed, Nov 26, 2008 at 11:30 AM, henkoegema <[EMAIL PROTECTED]>wrote:
>
>
> Anthony Minessale-2 wrote:
> >
> > 1) Open your browser to the bounty page
> > 2) Post a bounty to add that feature.
> >
>
> What (where) is the bounty page ?
> --
> Vie
On Wed, Nov 26, 2008 at 6:14 AM, Brian West <[EMAIL PROTECTED]> wrote:
> For now I only want it to set one domain. Because we are creeping
> outside the scope of what the default config can do. You really
> should create your own context for inbound calls and set the
> domain_name before you tra
Check out the various APIs that FS can support:
http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC
http://wiki.freeswitch.org/wiki/Event_Socket
The event socket is awesome and you only need a socket-level connection to
control FS and you can do it from any machine with connectivity to the FS
serve
Why, yes there is! You want the "originate" command:
http://wiki.freeswitch.org/wiki/Mod_commands#originate
I'm not sure I understand your API command question, however you can
definitely execute originate commands via the event socket:
http://wiki.freeswitch.org/wiki/Event_Socket
Could you give
FYI,
I've updated the wiki to reflect the current status of OpenMRCP with a link
to the new UniMRCP project. Hopefully enough people who want MRCP in FS will
support UniMRCP...
-MC
On Mon, Dec 1, 2008 at 11:17 AM, Anthony Minessale <
[EMAIL PROTECTED]> wrote:
> mod_openmrcp was a contribution to
Bring on SNAP, baby!
On Tue, Dec 2, 2008 at 11:03 AM, Kristian Kielhofner <
[EMAIL PROTECTED]> wrote:
> On 12/2/08, Anthony Minessale <[EMAIL PROTECTED]> wrote:
> > Naturally, either way is stupid.
>
> Word.
>
> > The whole idea of putting the transport in a uri param is equally stupid
> to
> >
Right now this page is up-to-date with the latest info:
http://wiki.freeswitch.org/wiki/Mod_fax
T.38 is not (yet) supported.
-MC
On Tue, Dec 2, 2008 at 9:40 AM, Dennis <[EMAIL PROTECTED]> wrote:
> hi,
>
> because we do not get tired of testing and playing a lot with the
> beloved fs, we now arr
On Tue, Dec 2, 2008 at 1:28 PM, Kristian Kielhofner <
[EMAIL PROTECTED]> wrote:
> On Tue, Dec 2, 2008 at 3:32 PM, Michael Collins <[EMAIL PROTECTED]>
> wrote:
> > Right now this page is up-to-date with the latest info:
> > http://wiki.freeswitch.org/wiki/Mo
Kristian,
Are you on the IRC channel by any chance?
-MC (IRC: mercutioviz)
On Tue, Dec 2, 2008 at 1:28 PM, Kristian Kielhofner <
[EMAIL PROTECTED]> wrote:
> On Tue, Dec 2, 2008 at 3:32 PM, Michael Collins <[EMAIL PROTECTED]>
> wrote:
> > Right now this page is up-to-d
hehe, careful what you wish for...
On Tue, Dec 2, 2008 at 5:11 PM, ccav <[EMAIL PROTECTED]> wrote:
>
> RESOLVED.
>
> Duh, I'm sposed to use ringback, not playback...
>
> Someone should write a book on this... Maybe I will.
> --
> View this message in context:
> http://www.nabble.com/Wrong---in-v
You probably have several options depending upon your needs. Could you
elaborate a bit on what the big picture is? Also, what exactly were you
doing when you established the second call leg? Did the second call let get
created and a valid uuid assigned, etc.? Just checking.
Let us know,
MC
On Tue
On Wed, Dec 3, 2008 at 10:27 AM, Lachezar Valchev <
[EMAIL PROTECTED]> wrote:
> Hello everybody,
>
> I am new to the list and I hope I can find some help here, regarding an
> issue I am experiencing with the CDRs written by Freeswitch.
>
> The thing is, I am using the "max-sessions" and the "sessi
On Wed, Dec 3, 2008 at 10:27 AM, Lachezar Valchev <
[EMAIL PROTECTED]> wrote:
> Hello everybody,
>
> I am new to the list and I hope I can find some help here, regarding an
> issue I am experiencing with the CDRs written by Freeswitch.
>
> The thing is, I am using the "max-sessions" and the "sessi
Hi Gab!
Welcome to FreeSWITCH. Thanks for your questions. I'm trying to learn all of
this stuff and help others so bear with me while I research these and help
you find the answers.
BTW, are you on IRC? you can visit us for realtime help, #freeswitch on
irc.freenode.net
-MC (mercutioviz on irc)
And thank you for testing and being gracious! :)
-MC
On Wed, Dec 3, 2008 at 1:24 PM, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> I tested both patches from the trunk : network_addr is set to the remote
> IP on the b-leg
> and local media port and remote media port hold the correct values when
Nice! I'll add that to my list of new prompts to be recorded. FYI, if
you have any other suggestions please email this list or post a
comment here:
http://jira.freeswitch.org/browse/FSSCRIPTS-9
-MC
On Thu, Dec 4, 2008 at 9:25 AM, Gabriel Kuri <[EMAIL PROTECTED]> wrote:
>
> How about:
>
> "The mai
I think GM Voices levies a "naughtiness surcharge" but I'll see what I
can find out. :)
-MC
On Thu, Dec 4, 2008 at 9:50 AM, Anthony Minessale
<[EMAIL PROTECTED]> wrote:
> maybe
> http://www.sofaswitch.org/eg/sounds/fucked.wav
>
>
>
> On Thu, Dec 4, 20
Dennis,
Thanks for your input on the fax stuff! We will check this out and report back.
Question: if one of the devs would like to SSH into your system to do
further testing, is that okay?
Thanks,
MC
On Thu, Dec 4, 2008 at 11:45 AM, Dennis <[EMAIL PROTECTED]> wrote:
> hi,
>
> after we managed t
Check it out:
http://digg.com/software/FreeSWITCH_knocks_Asterisk_s_block_off
Please diggit left and right!!
-MC
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Is this for Music on Hold? Or is it for a different application altogether?
Thanks,
MC
On Fri, Dec 5, 2008 at 12:37 AM, Faisal Maqsoodi
<[EMAIL PROTECTED]> wrote:
> Hi,
> Can i accomplish folder tasks with freeswitch? For instance, i need to play
> all sound files contained in a directory sequent
On Fri, Dec 5, 2008 at 8:32 AM, Frank @ Impact <[EMAIL PROTECTED]> wrote:
> Looks like tone detect might do it. But..
>
> If so, What frequency would we use for particular keys?
>
http://en.wikipedia.org/wiki/DTMF#Keypad
> Will tone_Detect sniff both legs or would we just do both r and w on the
Peter,
thanks, I will ruminate on this and get back with you as soon as I can.
-MC
On Fri, Dec 5, 2008 at 9:08 AM, Peter P GMX <[EMAIL PROTECTED]> wrote:
> I am a step further, When I set the cid-name then I can access the data
> dring
> channel_outgoing
> channel_originate
> channel_progress
> c
On Fri, Dec 5, 2008 at 8:41 AM, Anthony Minessale
<[EMAIL PROTECTED]> wrote:
> call_timeout is only used if you are bridging 2 channels where one or both
> of them is still unanswered.
>
> what you want to use is originate_timeout and forget about call_timeout
>
> you also have
> leg_timeout and le
Doh! Brian is way ahead of me, as usual...
On Fri, Dec 5, 2008 at 11:05 AM, Brian West <[EMAIL PROTECTED]> wrote:
> But you don't see the invite hitting FreeSWITCH? And you're behind
> NAT? Make it register every 30 seconds instead of the default 3600
>
> /b
>
> On Dec 5, 2008, at 10:59 AM, MEHD
Can you hit F8 and capture the debug output when making a call?
That'll help us see what's going on.
-MC
On Fri, Dec 5, 2008 at 10:59 AM, MEHDi CHAABOUNi
<[EMAIL PROTECTED]> wrote:
> Actually, i did not mean that the line is dropped during a call...
> FS is configured to accept calls from the Junc
Joe,
A few questions... what svn rev are you running? Which operating
system? Finally, is it possible for you to put your dialplan and Lua
script up at pastebin.freeswitch.org?
Thanks,
MC
On Mon, Dec 8, 2008 at 8:57 AM, Joe Bain <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I'm writing an IVR in Lua and
Thanks for your feedback. It definitely helps to know not only what
you need FS to do but why you need it to do so.
Do you have FS in production right now? Just curious.
Thanks,
MC
On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos
<[EMAIL PROTECTED]> wrote:
> "I already added 2 patches fo
Which OS are you running?
-MC
On Tue, Dec 9, 2008 at 2:07 AM, Helmut Kuper <[EMAIL PROTECTED]> wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hello,
>
> I tried to compile mod_fax today with trunk from yesterday. A 'make' in
> FS trunk directory led to an error saying that libspandsp
Thanks again for the heads up. We'll check it out.
-MC
On Tue, Dec 9, 2008 at 2:12 AM, Helmut Kuper <[EMAIL PROTECTED]> wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hello,
>
> I tried compile FS with mod_xml_ldap with trunk of yesterday. During
> compiling it can't find
> http://sv
Helmut,
I think Mike J was pointing out that spandsp needs libtiff and
libtiff-devel in order to compile, so you need to do that first and
then compile freeswitch.
-MC
On Tue, Dec 9, 2008 at 8:09 AM, Helmut Kuper <[EMAIL PROTECTED]> wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi M
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