bet for someone
new to the software?
Thanks
Tim
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Michael Collins
Sent: Saturday, April 05, 2008 3:04 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] freeswitch as a SIP Bridge
Tim,
I can
-0700, Michael Collins wrote:
Tim,
If you're comfortable with Linux then I would suggest starting
there,
but you are by no means locked in. Much of the development of FS is
done in CentOS, and I believe in a 64-bit hardware environment.
You'll also find that the current
I know IRC user jdfiles is/was at one point using an A101 as a PRI GW
for his SIPx PBX. I'll forward your email to him and see if he's still
using it...
-MC
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Leonardo Alves
Sent: Wednesday,
I haven't updated in a while. Are there any recent changes I should know
about?
Jeff,
The PRI stack has seen some improvements since we last spoke. I highly
recommend updating if only to get the latest OpenZAP stuff.
Thanks for the update!
-MC/mercutioviz
Brian,
Your example is extremely well-presented! I can't think of anything off
the top of my head that we could/should add at the moment, but if you
guys are on IRC and you keep getting certain dialplan-related questions
then shoot an email to the -users list with a suggestion for a dp recipe
Chris,
Thanks for staying on top of wiki changes!
-MC
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Chris Danielson
Sent: Tuesday, April 15, 2008 12:43 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] How to
Brian,
I think this is a great idea. 1PM CST works for me, and the middle days
of the week are best, as others have pointed out. Tu-W-Th are all good
choices. For me personally, Thursday is the absolute best day because
I'm frequently out of the office and have more flexibility in my
schedule
I'd like to try one tomorrow. I don't know if I can stay on for two
solid hours but I will give it my best shot.
-MC
-Original Message-
From: [EMAIL PROTECTED]
[mailto:freeswitch-dev-
[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Tuesday, April 29, 2008 11:22 AM
To:
I scheduled it for 2 hours but for most of them they'll be 1 hour..
but that should give us some spare time.
Can you pick 10-20 API's and Apps that we can go over and verify all
the docs on the wiki...
I'll throw in a few...
It seems there've been some recent threads about how to handle
FYI,
Just thought I'd share. I spoke with someone in my area who had heard
of FS. This is the first time I've run into someone in person who had
heard of FS but who had not heard of it from me. FYI, the guy is a
former ATT sales rep and is now working for Telepacific. He said that
he's got
Tony has a valid point here. Is there any way for you to try on
different hardware? Portaudio, in the grand scheme of things, isn't
very high on the food chain when it comes to FS development, especially
with 1.0 only a few weeks away. Please see if you can reproduce these
symptoms on other
Followup question: I can use api to send show channels or oz list
but when I use bgapi to send those commands I don't get anything back
from the event socket. Any thoughts?
-MC
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Michael Collins
bgapi show channels
but, you can also say bgapi show channels from the
cli/xml-rpc/dialplan etc as well.
On Fri, May 16, 2008 at 1:08 AM, Michael Collins
mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Followup question: I can use
@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Event socket: api vs. bgapi
yes that module is out of date.
see the FreeSWITCH::Client in tree for mine
On Fri, May 16, 2008 at 1:09 PM, Michael Collins
[EMAIL PROTECTED] wrote:
Yep, that seems to work. I believe the issue is with
POE::Filter
PHP?! Yucky! :-(
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
EdPimentl
Sent: Friday, May 23, 2008 8:23 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] XML based IVR
I would like to see an XML IVR
FYI,
I've been playing with this and it is very cool. I am working on a mini
web-page that will be like a control panel, although it's just for me to
watch what's happening on my FS box via a web page using Dojo widgets +
Ajax/(D)HTML/CSS. I have a need for my users to be able to monitor
Ward,
Any chance we'll see some of your sweet recipes on nerdvittles with a
little FS baked in? :D
-MC
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ward
Mundy
Sent: Tuesday, May 27, 2008 4:53 PM
To:
: [Freeswitch-users] RESTful Bounty . Putting the
competitionto REST with RestFul VoIP2.x services
On Tue, May 27, 2008, Michael Collins wrote:
Re: Any chance we'll see some of your sweet recipes on nerdvittles
with a little FS baked in? :D
You bet. In fact, we're raising a bounty
Subject: Re: [Freeswitch-users] RESTful Bounty . Putting the
competitionto REST with RestFul VoIP2.x services
Oh hell no
From: Michael Collins [EMAIL PROTECTED]
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Wed, 28 May 2008 09:52:21 -0700
of usable AMD here in the US (and no app_amd is not
legally usable in 90%+ areas of the US that's why it was released free
like that... Most areas have a 2sec media rule and app_amd violates the
crap out of that)
K
From: Michael Collins [EMAIL PROTECTED]
Reply
Gang,
I'm working on my outbound IVR and I'm wondering if you have any
thoughts on this topic: I'd like to make a record of the digits dialed
by the called party. For example, I have an IVR menu that says, to
repeat these options, please press star. I'd like to record how many
times the
FYI,
I create a new wiki page:
http://wiki.freeswitch.org/wiki/Rosetta_stone
I thought maybe those who know about Asterisk and FreeSWITCH both could
start dropping stuff in there. I think that people coming from Asterisk
might have an easier time if they had some sort of frame of
Anthm/MikeJ/bkw,
Any chance you could write a few paragraphs about media bugs? Nothing super
in depth, just a basic definition of what they are, why one might use them,
etc. I'd like to update the wiki with at least a basic write-up on media
bugs.
Thanks!
-MC
Guys,
I don't know if this is normal or not so I'm hoping you can help me
figure out what's up. I've got a disconnected number that I call on a
PRI and the sequence goes like this:
Dial number, hear ring back, see PROGRESS from telco, hear SIT and
you've reached a DC'd number..., then
Are you leaning in any direction in terms of building the GUI? JS
libraries like Prototype, Dojo, etc.? Web framework? Back end
scripting lang? I'm just curious what that decision making process
looks like and what you've thought about thus far.
-MC
-Original Message-
From: [EMAIL
By the way this was one of the few things that I had already put on my
work-in-progress Rosetta stone wiki page:
http://wiki.freeswitch.org/wiki/Rosetta_stone#Miscellaneous
-MC
-Original Message-
From: [EMAIL PROTECTED]
[mailto:freeswitch-
[EMAIL PROTECTED] On Behalf Of andrew
Sent:
BTW, if you launch it from that directory it will find the fs::client
package, but if you move that script to a more permanent place, like
/usr/local/freeswitch/scripts then you'll need to make sure that
fs::client stuff bkw mentioned is installed in your local Perl or is
otherwise accessible to
Unfortunately there isn't an equivalent of show dialplan in FS. There
is a show channels. In fact, if you connect to the CLI with the perl
script mentioned below then you can just type show and you'll see all
sorts of stuff that you can do. Another fun thing is to enable xml-rpc
and then you
Very interesting! I had heard of VoiceXML but not CCXML. I will
definitely check it out. I think the big question would be: is it worth
it (yet) to add a CCXML interpreter to FS? I suppose another question
would be: are there any CCXML utilities, parsers, libraries, etc. that
could be
Brian,
Is this syntax going to be tagged for 1.0.1? I ask because the
documentation guys are trying to get a handle on the deltas between
1.0.0 and 1.0.1.
Thanks!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:freeswitch-
[EMAIL PROTECTED] On Behalf Of Brian West
Sent:
Sandeep,
In addition to the stuff Brian and Anthony asked you about, please do
these steps:
#1 - if you didn't make any modifications to your
conf/dialplan/default.xml file then delete it
#2 - in the fs source directory do make current which will update
everything
#3 - when the update is
Check mod_shout.c
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of e
schmidbauer
Sent: Tuesday, June 24, 2008 11:37 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_shout.c:497 write_stream_thread()
MP3encode error
You said: If XML is the single reason you won't be using FreeSWITCH
then by all means stick with Asterisk, because we aren't going to beg
you to use FreeSWITCH.
This is sad, you claim that FreeSWITCH is more advanced in terms of
features and innovation than Asterisk, but all I see with your
Sorry I've been slacking, but I finally got this one wikified:
http://wiki.freeswitch.org/wiki/Mod_fifo#Agent.2FCaller_Example
-MC
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Anthony Minessale
Sent: Wednesday, July 02, 2008 9:03 AM
To:
FYI,
You know the drill. Here's the link for the agenda for tomorrow:
http://wiki.freeswitch.org/wiki/Wiki_meet_2008_07_09
Please add whatever you think is relevant for tomorrow.
-MC
___
Freeswitch-users mailing list
Please also indicate just how much you'd like it, i.e. if you are
willing to contribute to the devs for their time and effort.
-MC
-Original Message-
From: [EMAIL PROTECTED]
[mailto:freeswitch-dev-
[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Tuesday, July 08, 2008 3:23 PM
To:
We would support this.
Teliax has been doing some testing on FS and would need support for a
full fledged, scalable IAX channel to go any further. I think anthm
has the scope of the project in mind. If the community is interested
we'll gladly contribute $$$ as well as dev support on an
I am experimenting with XML CDRs and I was curious about this part of
the record. It seems that with each transfer to a different extension
I'm getting a new callflow profile. That's good - I just want to make
sure that my understanding of this is correct.
Will a new callflow profile get
Followup - wait about good ol' fashioned silence detection, akin to
WaitForSilence in ast?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:freeswitch-
[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Wednesday, July 09, 2008 2:12 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] silence, non-silence or tone detection
look again =D
On Wed, Jul 9, 2008 at 4:56 PM, Michael Jerris [EMAIL PROTECTED] wrote:
We don't have one in tree, but it wouldn't be hard to write one.
Mike
On Jul 9, 2008, at 5:20 PM, Michael Collins wrote:
Followup
Here's the new link:
http://wiki.freeswitch.org/wiki/Wiki_meet_2008_07_16
it's pretty bare right now so add some items for discussion tomorrow.
I'm sure that the wiki is not perfect! :-)
-MC
___
Freeswitch-users mailing list
Is there any reason that the uuid field doesn't work for you?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Ashutosh
Sent: Wednesday, July 23, 2008 12:18 PM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Equivalent of
Guys,
I was just looking over the source directory an I noticed that the
common files for an open source project may need some serious attention:
AUTHORS
INSTALL
NEWS
README
I'm sure that the COPYING file is just fine.
NEWS and README are completely empty. Should we maybe just put
Recently a bug has been opened on the default config because the
context on all the profiles are set to public. Let me take a few
moments to clarify WHY its like this.
Hehe, security as a forethought should never be considered a bug!
The internal profile is the one setup in the default
Here's the agenda:
http://wiki.freeswitch.org/wiki/Wiki_meet_2008_07_30
All are welcome.
-MC
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Speaking of the Rosetta Stone page, it could use some love and
attention. If you have any Asterisk and FS knowledge we'd love to have
you add something to the Rosetta Stone page. Even if you add just one
tip or trick it would be welcomed heartily. It could save a lot of time
and energy on the
Hi Mike,
Any chance of you posting that information?
If you did so already, sorry, I might have missed out on your posting.
Thanks, Birgit
:(
I am a slacker. Stand by and I will get an XML-CDR wiki page started
and I'll get these vars documented first.
Please give me an hour or so and
not have editing collisions.
:)
-MC
On Thu, Jul 31, 2008 at 11:11 AM, Michael Collins [EMAIL PROTECTED]wrote:
Hi Mike,
Any chance of you posting that information?
If you did so already, sorry, I might have missed out on your posting.
Thanks, Birgit
:(
I am a slacker. Stand
I think you meant this command:
http://fs.ip:8080/webapi/sofia?status%20profile%20internal
Sorry, and yes. :) The issue also is apparent when doing the version
api:
http://fs.ip:8080/api/version yields nothing
http://fs.ip:8080/webapi/version yields the correct information
In any case I just
commands like show and status can tell they are
being accessed over the web and act accordingly with /api /txtapi/ and
/webapi/
On Fri, Aug 1, 2008 at 5:05 AM, Michael Collins [EMAIL PROTECTED]
wrote:
I think you meant this command:
http://fs.ip:8080/webapi/sofia?status%20profile%20internal
Simon,
Yes to your first question and I think so to your second. See
conf/autoload_configs/console.conf.xml. there are some notes in there about
how to configure which pieces get logged and for which log levels.
-MC
P.S. - I'm off to cluecon so I'll check in later...
On Mon, Aug 4, 2008 at
As a token of your appreciation you could add this to the Rosetta Stone
page... :D
BTW, any time someone says, Here's how we do it in Asterisk and then
someone else says Well in FS you can do that like this... it would be
great to add that to the Rosetta Stone page. That page is kinda thin and
I just grepped the entire source tree and toll_allow is not a predefined
chan var. In other words, Brian just demonstrated the power and flexibility
of FS by giving you just a few lines of config.
setting toll_allow to a comma-delimited value lets you have have multiple
values that can be
That begs the question... is there a mechanism in sqlite or Linux that
allows for the RAM drive to be backed up periodically? That would be a
cool feature to get documented for those power users like Ken! ;)
-MC
From: [EMAIL PROTECTED]
[mailto:[EMAIL
I have one of those. :-)
Actually, its a bit more than you said. You need to cater for wink
start
and a few other things. Nothing too difficult, though.
Steve
Steve,
As usual you are awesome! Is there anything the RBS users can do to
grease the wheels? Maybe we can do a reasonable bounty
Here's the current link:
http://wiki.freeswitch.org/wiki/Wiki_meet_2008_08_20
Please add your questions/concerns/comments, etc.
-MC
___
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Freeswitch-users@lists.freeswitch.org
A couple of options but something like a NxtVox card would give some
flexibility. You could get the NxtVox NXA400P with one FXO port module
and then if you want to get crazy later you could add FXS ports, etc.
Or if you want to go really cheap (and *really* scary) you could look
for a X100P clone
Well, I'm 3 hours away but I'd still like to come visit every now and
again. I'll just drop the wife kids at a relative's house in SJ and
we'll go from there! :-)
-MC
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Darren Schreiber
Sent:
Tomorrow's documentation meeting:
http://wiki.freeswitch.org/wiki/Wiki_meet_2008_08_27
Please add items as you see fit.
-MC
___
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Freeswitch-users@lists.freeswitch.org
Start here:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_read
There are a few examples. I don't know if this is the app you need for
production but it will definitely let you read a DTMF and thus is sufficient
for testing whether or not your system is seeing DTMFs from callers.
FYI,
Here's your link:
http://wiki.freeswitch.org/wiki/Wiki_meet_2008_09_03
Thanks to lanman for getting this ready. FYI, I will not be on the call
tomorrow because I will be at the orthodontist... I will check in on
Thursday to see how it all went.
-MC
Luis,
Can you re-post your dialplan extension? Also, need to make sure that
you don't have ignore_early_media=true because that will throw you off.
-MC
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Luis
F Urrea
Sent: Tuesday, September
Here's a snippet of what I use with tone_detect, although I don't
actually look for a ring signal:
extension name=ivr_start
condition field=destination_number expression=^82(\d{10})$
!-- we can skip all this early media stuff if the far end answers
--
action
OpenBTS is GPLv3... may have OS licensing issues trying to make that happen.
-MC
On Fri, Sep 5, 2008 at 11:56 PM, Tamas [EMAIL PROTECTED] wrote:
Hi,
Have you seen this?
http://openbts.sourceforge.net/background.html
I guess, FreeSWITCH would be better for this ;)
Regards,
Tamas
Folks,
I got two questions:
1) How do i configure Freeswitch to detect inband DTMF?
Start here:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf
2) How do i play DTMF to the remote caller?
Check this out:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_send_dtmf
Maybe,
But, I achieved it this way:
[EMAIL PROTECTED] originate A B
[EMAIL PROTECTED] show channels (find out corresponding UUIDs)
[EMAIL PROTECTED] uuid_transfer UUID of A or B C
If you haven't already figured it out then remember this axiom:
In FreeSWITCH, if you know the UUID of
You don't setup gateways to receive calls. You can just have your
provider send as many did's at you as they feel free to send. Just
don't auth-calls on that profile that all the did's hit. A gateway has
nothing to do with DID's it just registers to the far side and is used
when they
IIRC, Anthony mentioned an elegant way of handling this not too long
ago, although it was with a different language. The gist of it was this:
While (session.is.active) {
Do stuff
}
Do stuff after hangup
Obviously you'll need to use Python-ish syntax but you get the idea.
Alternatively you can
Evgeniy,
I've only got a few minutes so I have to be quick.
Right now we only support the user or TE (terminal) side of PRI. Still
working on the network side. If you need to do a back-to-back setup for
testing then your options are a bit limited, but I do know that YATE and
Asterisk both have
Evgeniy,
I'm not sure why EuroISDN isn't listed in the enum or string below because I
know that Stefan (stkn on IRC) has been working on it for some time. I'll
follow up with him and Mike Jerris to see what's going on there.
As for seeing the dialect at run-time, right now you can't, although I
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 PROTECTED]
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 mailing list
Freeswitch-users@lists.freeswitch.org
Jon,
Are you saying that you cannot pass a comma-separated list of variables
like this?
action application=bridge
data=[origination_caller_id_number=45161061,my_var1=blah,my_other_var=w
hatever]sofia/gateway/45161061/$1/
Note: no spaces!
-MC
From: [EMAIL
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, September 23, 2008 12:58 PM
To: freeswitch-users
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:
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
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
Santos
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Michael Collins
Sent: Wednesday, September 24, 2008 4:17 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Voicemail x DID
Jair,
Are you meaning
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 transfer with data
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 requited
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
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
wiki
, 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 post your
PHP code
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
param name=apply_nat_acl value=rfc918/
param name=agressive_nat_detection value=true/
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
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 of
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 list
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-users]
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:
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
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 -
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:
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 and
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_Management
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
: 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 want
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
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
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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 one
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