I highly recommend that you set aside this endeavor for the time being and
use the default configuration. Once you get familiar with the default config
then you'll realize how to make changes to registered users and to the
dialplan. Don't let the size of the default configuration scare you off. It
Are both of these phones on the same LAN as FreeSWITCH? What kind of phones
are they? Also, can you reverse the dialing order and reproduce the
symptoms? Just curious if it's always the same phone causing the noise or if
it is always the second phone connected, regardless of which phone.
-MC
On M
Hey, this would be great info to put on the wiki... (hint hint wink wink
nudge nudge) :)
-MC
On Thu, Apr 9, 2009 at 8:51 AM, Chris Fowler wrote:
> I'm running FS on Amazons' EC2 compute cloud (AWS) and have 30 Polycom
> phones working happily in this config.
>
> I modified the Internal profile i
John,
Just curious - why are you using zaptel at all? Does it provide something
for you that the wanpipe drivers do not? I use Sangoma only with Sangoma
cards and I have a lot of success.
-MC
On Wed, Apr 8, 2009 at 3:19 PM, John Wehle wrote:
> > Okay, a few things. First off, the wanpipe2.conf
John,
Okay, a few things. First off, the wanpipe2.conf file has a booboo. This
line is WRONG:
TDMV_DCHAN = 0
For ISDN in North America you want:
TDMV_DCHAN = 24
Also, I recommend changing this line:
wbg1 = wanpipe2, , TDM_VOICE, Comment
To this:
wbg1 = wanpipe2, , TDM_VOICE_API, Comme
>
>
> 2009-04-08 00:43:24 [ERR] mod_sndfile.c:194 sndfile_file_open() Error
> Opening File
> [/usr/local/freeswitch/sounds/en/us/callie/conference/conf-pin.wav]
> [System error : No such file or directory.]
> 2009-04-08 00:43:24 [WARNING] mod_conference.c:4799
> conference_function() Cannot ask the
On Fri, Apr 3, 2009 at 7:11 AM, Brian West wrote:
> Did it sound more like a machine gun?
> /b
>
>
Comfort noise for General Douglas McArthur I guess...
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FYI, these are good questions but they probably belong on the dev list since
they are so technical in nature. :)
-MC
On Fri, Apr 3, 2009 at 6:20 AM, Lele Forzani wrote:
>
> Hello,
> I've been experimenting with the use of mod_dahdi_codec and other ways
> to perform external transcoding for codec
Thanks for doing some of the legwork on this. BTW, this thread is probably a
bit too technical for the users list - I recommend sending to the dev list.
:)
-MC
On Thu, Apr 2, 2009 at 9:46 AM, Tamas Cseke wrote:
> Hello,
>
> We originate loopback channels and they end up in calling sofia
> and tr
That being the case, maybe a pcap of the audio might yield some clues?
On Wed, Apr 1, 2009 at 3:34 PM, Thorhallur Sverrisson wrote:
> The Polycom 650 is an IP phone, so the ground loop should not apply.
> Ground loops occur only in analog systems.
>
> As to what the buzzing is, I'm not sure. I h
Rupa,
Thanks for adding to the project! Well done.
-MC
2009/4/1 Rupa Schomaker
> Announcing a new module: mod_memcache
>
> Up until now one had two choices for storing arbitrary key/value pairs.
> hash or db. hash is fast, but it is local to the current FreeSWITCH
> instance. If you run multi
2009/4/1 Stromin Normin
> Hi,
>
> I've been asked to do some testing on Freeswitch by work, we currently use
> Asterisk. I'm quite new to telephony so please go easy.
>
> I have FS setup on a windows box and at the moment I'm testing internal
> calls only, when I transfer calls or call extensio
On Wed, Apr 1, 2009 at 11:46 AM, Tim Ringenbach wrote:
> Anthony Minessale wrote:
> > Did you follow the link I posted?
> > http://www.google.com/search?q=janitor+project
> >
> > The linux kernel calls it the same thing and so do all the other
> > project that come up in that search.
> >
> > Woul
2009/4/1 Carlos Talbot
>
> Is there an interest in running FreeSWITCH on OpenWRT? I recently managed
> to compile and run a version for a MIPs based router, the Planex MZK-W04NU.
> This router has 32MB ram, 8MB flash, runs at 400MHz, draft N support
> (2.4GHZ), based on 2.6 kernels, a usb port an
2009/4/1 Even André Fiskvik
> You're one very fine janitor Michael!
>
How DARE you call me a janitor! :)
> On the topic of the Janitor Project, this is how it should be.
> Devs give user feature => user documents new feature/behaviour.
>
Thanks. This is totally reasonable. Power users and new
2009/4/1
> Pardon me, but you speak only for yourself. I think Janitor is not an
> appropriate word.
>
I *like* janitors. I *respect* janitors. They are *honorable* and *
hard-working*. In short, we need janitors - people who are willing to roll
up their sleeves and get work done. Let's agree
The FreeSWITCH team would like to let everyone know that the latest version
is available. More information can be found here:
http://www.freeswitch.org/node/172
By all means download, upgrade, test, and report back! Your feedback helps
us make FreeSWITCH even better!
-MC
__
2009/4/1 Anthony Minessale
> pin checks and lock checks are both intentionally skipped on outbound calls
> transferred back to the conference.
> The idea is if you purposely placed an outbound call that was intended to
> land in the conference
> you would not want to do so only to tell them it's
Dear FreeSWITCH Community:
As you know, FreeSWITCH has been growing leaps and bounds and it's going to
keep growing as the word spreads. The core development team of Anthony,
Mike, and Brian are very appreciative of the community's help and
involvement in the project. Simply put: the community is
2009/3/31 Raffaele P. Guidi
> I am a Yate user and I can tell their mailing list suffer the same problem.
> My solution? I often ask for help but, as a personal policy, I always write
> an article or add to an existing one on the project wiki explaining and
> documenting what people explained to
FYI, I've documented the msleep method here:
http://wiki.freeswitch.org/wiki/Mod_lua#freeswitch.msleep
I will work on better and more organized API documentation. If anyone out
there has time/energy/knowledge of the scripting APIs and is willing to help
out please email me off list.
-MC
2009/3/3
Are logging both a- and b-legs? Just curious what your setup is.
-MC
2009/3/30 Keith Laaks
> Hi,
>
>
>
> I have an application where my Javascript hanguphook code calculates a
> value (e.g. the cost of the call which can only be calculated post hangup)
> and I need to have that value appear as
On Mon, Mar 30, 2009 at 10:26 AM, Bipin Patel wrote:
> hi,
>
> i currently live in a country called UAE - united arab emirates and a
> city called Dubai.
>
Hehe, Dubai is quite a popular place - even a number of us ignorant
Americans have heard of it! :) We would love to see FreeSWITCH become mo
> living in this region and being able to test and try open source softwares
> is not easy coz never 2 guys meet and end up talking something about linux,
> pbx, softswitches etc coz no1 has ever heard of such things other than the
> stuff whats running in the economy.
>
Out of curiosity, what par
Just following up... did you get these questions ironed out?
-MC
On Wed, Mar 25, 2009 at 2:51 AM, Gilles wrote:
> Hello,
>
> I have a couple of questions related to having SIP users connecting
> from the Net to a Freeswitch server through NAT routers on both ends:
>
> 1. How must I configure rout
> con = freeswitch.EventConsumer("all");
>
> now you have a consumer obj
>
> every time you call con:pop() with no arg you will either get an event or
> nil when there are no events to consume.
> every time you call con:pop(1) the consumer object will block until there is
> an event.
>
> So you use
On Thu, Mar 26, 2009 at 4:09 PM, Peter P GMX wrote:
> Hello Michael,
>
> I tried this, but received the same behaviour. It does not ask for the
> defined PIN.
Just curious - where do you define the PIN for this conference?
-MC
___
Freeswitch-users mail
On Thu, Mar 26, 2009 at 1:50 PM, Peter P GMX wrote:
> Hello,
>
> when I originate a call via event socket and transfer it to a
> conference, it doesn't ask for a PIN.
> Example:
> originate sofia/default/222331 &conference(222500)
> The same happens when I originate a call and transfer it to the 2
Look in the default.xml dialplan file for the "tod_example" extension.
(It's near the top of the file.) It has an example of how to create an
extension that simply sets a variable that can be used in other
extensions in the dialplan.
-MC
2009/3/26 Rodrigo P. Telles :
> Hi Guys,
>
> I'm trying to
Hey, if you guys get this all figured out, tested, and working then
please be sure to put it on the wiki. You could create a whole new
page and then link to/from the mod_conference page.
-MC
On Wed, Mar 25, 2009 at 7:02 AM, Szymon Olko wrote:
> Steven Ward pisze:
>> Szymon, I want to provide a se
2009/3/25 :
> Thanks MC, maybe a link to that "TortoiseSVN" would help for some in the
> Windows crowd.
Done!
>
> TortoiseSVN has a bunch of stuff in it but to make it simple, especially for
> doing updates to report bugs, then mentioning if doing just an "SVN update"
> will work before rebuild.
> Maybe the reporting bugs wiki needs updating for Windows users (experienced
> and inexperienced).
Quite possibly. The instructions are not explicit. I will add
something for the Windows users that's a bit more specific.
-MC
___
Freeswitch-users mailin
On Wed, Mar 25, 2009 at 8:43 AM, Pablo Hernan Saro wrote:
> First of all, thanks for your answers. You guys are awesome and FS
> rocks. Don't take it the wrong way...
No offense taken.
> My custom is to install production systems using stable versions and
> apply the corresponding security patch
2009/3/25 Matthew Fong :
> I'm wondering if there's any features that allow the cron-like execution of
> code inside of Freeswitch, preferably with lua--or if I am stuck using the
> api interface and running the cron outside of freeswitch.
> --matt
I guess the most important question you can answe
The FreeSWITCH Team is pleased to announce that all may register for
ClueCon 2009 immediately!
ClueCon is the Telephony Developers Conference by developers, for
developers. This year's event will be held at the beautiful Wyndham
Hotel in Chicago, August 4-6. Special room rates have been secured an
2009/3/24 Mitul Limbani :
> Brian,
> I can help with website, wiki and testing, tell me what's next step forward.
>
Mitul, do you have any experience with MediaWiki?
-MC
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> While FS official site has plenty of documentation, obviously we don't
> want to translate word by word. Then where should we start from? A
> forum?
I would start by finding as many people as possible who are literate
in both English and Chinese and who are willing to help out. Once you
have yo
On Mon, Mar 23, 2009 at 4:19 PM, Pablo Hernan Saro wrote:
> BTW, you also recommend SVN trunk for production servers?
> IMHO, should be a stable release for this purpose.
FreeSWITCH is one of those unusual projects where the SVN trunk is
generally more stable than the tagged releases.
-MC
__
On Thu, Mar 19, 2009 at 4:54 AM, Gilles wrote:
> Michael Jerris > There is currently no openzap (sangoma, etc) support
> on windows, we hope this will be coming soon.
>
> I found an alternative: The Linksys 3102 VoIP gateway. It's cheaper too.
>
> Would you say the Windows port of Freeswitch is r
On Thu, Mar 19, 2009 at 2:08 AM, Mark Tabron
wrote:
> So the second issue is possibly known - really could do with a fix or a
> workaround for this as we plan to use E1's for all incoming traffic.
>
> Can anyone shed light on the first problem (extension rings for a
> fraction of a second then han
> tone_detect! sounds good.
>
> BTW, was there any errors in those extensions I posted. I modified something
> you posted MC.
Not at first glance. What did you change?
-MC
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> From: freeswitch-users-boun...@lists.freeswitch.org
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
> Michael Collins
> Sent: 18 March 2009 17:24
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Is mod vmd working?
>
> 2009/3/1
On Wed, Mar 18, 2009 at 10:18 AM, Peter P GMX wrote:
> 2 months ago when I struggled with E1 trunks and OpenZAP in freeswitch
> there was a timer problem which was not solved yet. This caused channels
> to be busy in my case.
>
> I am not sure whether this is solved yet. Can anybody confirm?
>
We'
2009/3/18 :
> I added a voicemail tag in to a default extension 1001, I hear the
> voicemail beep but still don't see vmd_detect.
>
> Mark
FYI, I've used mod_vmd but only in a TDM environment on outbound calls
via a PRI. It worked very well on for detecting answering machine
beeps and vm be
> I'm afraid that your original bald claim - that "IVRs badly need echo
> cancellation" is simply
> wrong, misleading and irresponsible: those believing it will end up spending
> large sums
> of money on technology which they probably do not need.
Anybody with years, perhaps decades, of DSP progra
On Tue, Mar 17, 2009 at 3:35 PM, Mark Thomas
wrote:
> Hello, everyone.
>
> I am new to Freeswitch, and telephony in general. I am trying to set up a
> Freeswitch system at work for a project, and I have hit a wall.
>
> I have a dedicated LD T1 from Qwest and a Sangoma A104 card. I believe I
>
> any idea how i can fix this error ?
>
I believe this is a harmless warning. However, you might try to use
ozmod_libpri, which uses the libpri PRI stack instead of the built-in
OpenZAP PRI stack. More info here:
http://wiki.freeswitch.org/wiki/OpenZAP#OpenZAP_Installation
-MC
__
On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron
wrote:
> Another update - this time (part) good news! Decided to run wancfg_tdmapi
> again, using the same settings as we always did, and we can now make external
> calls. I suspect that whatever BT did yesterday kicked the circuit back into
> life.
2009/3/16 Steven Ward :
> I simply moved the file defining the gateway to conf/sip_profiles/internal
>
> Well, when calling from extension 1000 to 70904, what I see on the console
> (debug mode) is:
>
> 2009-03-16 13:35:39 [DEBUG] switch_core_state_machine.c:152
> switch_core_standard_on_execute()
2009/3/16 Steven Ward :
> Yes, the obvious is the case. :) I don't want to do a STUN lookup - the two
> machines are on the same LAN.
>
> What's the best way to get the gateway to not do a STUN lookup? Do I need
> to disable STUN for the external
> profile or make this gateway use a different pro
hat appears when
> placing an outside call. Hopefully it can help to provide a possible answer!
>
> http://pastebin.freeswitch.org/7751
>
> Will setup an IRC client and see if I can log onto the channel.
>
> Thanks again!
>
> -Original Message-
> From: freeswitch-users-boun...@lis
> Is it possible to access FS DB to retrieve data? Where can i find details
> about that?
Could you be a little more specific? Also, on a standard Linux/Unix
install you could check here:
/usr/local/freeswitch/db/*db
-MC
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On Mon, Mar 16, 2009 at 8:29 AM, wrote:
> Hi,
>
> I have a (probably dumb) question that I just spent over 5 hours on:
> I have a 1.0.2 version running with 6 extensions, 2 SIP trunks and gtalk OK.
Ouch! Any way you could update? We are on the verge of releasing
1.0.4; 1.0.2 is OLD. :)
>
>
2009/3/16 Steven Ward :
> I'm trying to set up a sip trunk between a FS and * box, and right now I'm
> having trouble getting things set up so I make a call from a sip phone
> registered with my FS box to a sip phone registered w/ my Asterisk box.
>
> I have a gateway defined as in directory/defaul
FYI,
I've just started a new JIRA for the new voice prompts list that we
are getting together:
http://jira.freeswitch.org/browse/FSSCRIPTS-15
Please add comments to this JIRA if you can think of useful prompts
for FreeSWITCH that we don't already have recorded. We also need
financial support to g
On Thu, Mar 12, 2009 at 12:26 PM, wrote:
> Hello all,
> I'm new to this very nice system
> I'm looking into writing javascripts to interact with the system.
> How can one debug, run step by step and get variables values the
> javascripts running under this system? For example I have setup the sam
> I also double checked with wireshark and saw that the DTMF is SIP based, and
> the values were *3 and not 03 as FreeSWITCH reports.
> This probably is the problem and not the bind_meta_app.
> We are using bezeq international as our provider.
> DTMF are RFC2833.
> Thanks,
> Shahal
Shahal,
The d
> I have attached the freeswitch log http://pastebin.freeswitch.org/7730
>
> Can any one correct were i am wrong.
Baskar,
At first look your configs seem okay. Please pastebin the output of
these commands from the CLI:
oz list
oz dump 1
-MC
___
Freesw
> My first post to the list. I’m a bit of a newb to FreeSwitch (and linux) so
> apologies if some of my terminology isn’t quite correct.
Welcome to FS! Just out of curiosity, have you ever used Asterisk or YATE?
>
>
>
> Recently had a 9 channel ISDN30 (euro – q931) installed by BT (UK). We’ve
> h
> This is currently running trunk rev 12218 but I'm about to update to
> 12571 to see what happens.
To quote Samuel L. Jackson in "Jurassic Park":
Hold on to your butts!
-MC
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On Wed, Mar 11, 2009 at 10:42 AM, wrote:
> Mike, no luck with that either.
>
You may have to get fancy and use the transfer app and create an
extension like this:
Then just call it from the regular extension:
> I still need to see this through but another related approach w
On Wed, Mar 11, 2009 at 9:42 AM, wrote:
> I changed that tag so that hangup_after_bridge is false:
>
>
> but I still don't get the .js application working which is nothing more than
> a test script that ran if I dialed it's extension with the preceding one
> commented out:
Try adding a break="
On Wed, Mar 11, 2009 at 8:25 AM, Chris Danielson wrote:
> Hello Guys,
> I have a question regarding how I can do the following within the FS C API.
>
The devs love it when people get down and dirty with FS! However, this
thread is definitely better suited for the freeswitch-dev mailing
list. Also
> After I call the external number successfully, I’m able to receive DTMF from
> the softphone but the PSTN’s DTMF doesn’t work.
We definitely don't want to assume anything, so I have to ask the
obvious questions:
who is the provider?
are the DTMFs in-band or RFC2833?
Any chance you can turn on f
> I´m looking for suggestion on some features like:
>
> Accounting
> Rate
> Billing
> Route
There's nothing out-of-box ready, but you should look at these mods:
mod_nibblebill - real-time billing
mod_lcr - outbound routing
mod_easyroute - inbound routing
Start with the wiki: http:wiki.freeswitch.
> it works good. Can I assign a sleep to playback somehow to avoid loose
> of phrases at the beginning?
Are you playing just a single file? You can use the phrase macros to
create a pause at the beginning of your playback. Or you can cheat and
prepend a few hundred (or thousand) milliseconds of si
On Tue, Mar 10, 2009 at 6:16 AM, Anthony Minessale
wrote:
> Latest SVN:
>
> Send no extra caller id info:
> {sip_cid_type=none}sofia/default/u...@example.com
>
> Send RPID (default)
> {sip_cid_type=rpid}sofia/default/u...@example.com
>
> Send P-XXX-Identity
> {sip_cid_type=pid}sofia/default/u...@e
> [devices]
> wanpipe1 = WAN_AFT_TE1, Comment
>
> [interfaces]
> wp1aft1 = wanpipe1, auto, API, Comment
> wp1aft2 = wanpipe1, auto, API, Comment
>
> [wanpipe1]
> CARD_TYPE = AFT
> S514CPU = A
> CommPort = PRI
> AUTO_PCISLOT = NO
> PCISLOT = 4
> PCIBUS
On Fri, Mar 6, 2009 at 2:01 PM, Kristian Kielhofner
wrote:
> Why not just provide a kickstart file on freeswitch.org? It's pretty
> easy to pass them to the installer over the network and/or add them
> onto existing ISOs and other bootable media...
>
> I'd be happy to write/maintain such a thing.
On Fri, Mar 6, 2009 at 10:19 AM, Brian West wrote:
> The current default config comes out of the box as a 20 extension PBX config
> with various features including voicemail and conferencing.
> /b
And, unlike many other systems, you don't have to pay to increase the
number of extensions on the sy
> The FS log does not report anything abnormal. I'm running FS svn rev 11279.
> Does anyone know what could happened?
Well, you're about 1200 revs behind current SVN. A lot has been
improved in the past month or two, so definitely get yourself to the
latest SVN.
-MC
_
> Did anybody notice my email from yesterday that shows how there already is a
> forum on voip-info that is linked to our homepage and nobody uses it?
>
> We can't take this poll until we have a list of volunteers who would manage
> any new
> online resources.
Let's "end the torment of this thread
> Everyone seems to slate Centos, but to my surprise Anthony recommends Centos
> 5.2 which is nice to hear. Yes I know it’s not bleeding edge, but I don’t
> want that.
Repeat the mantra: CentOS is boring and predictable; boring and
predictable is perfect for real-time telephony systems.
> Any re
On Thu, Mar 5, 2009 at 12:20 PM, Peter P GMX wrote:
> Hello Brian,
>
> concerning
>> Well you should use ESL then ;)
> I simply do not understand what you mean by this. Is it sarcastic? Am I
> asking stupid questions?
>
ESL = Event Socket Library. It is an abstraction layer to make
interacting wi
On Mon, Mar 2, 2009 at 1:58 PM, Peter P GMX wrote:
> Hello Anthony,
>
> sorry for being tenacious but in some cases it works in a way we need it:
> If I a am not suppressing the cid numer when calling A, the following
> scenario works:
>
> * A receives a Call (originate) with CID '00' (
On Mon, Mar 2, 2009 at 11:48 AM, Anthony Minessale
wrote:
> i think that's what mod_vmd does
>
I think that's right. It just does the opposite - instead of looking
for differing power levels it looks for the same power level. In other
words it tries to detect distinctly non-human sound. I'll bet y
On Fri, Feb 27, 2009 at 6:07 AM, Helmut Kuper wrote:
> Hello,
>
> I play around with record_session and would like to have caller and
> callee separated on left and right channel. I found record_stereo is
> used for this. Unfortunately it doesn't work. A and B leg are still
> mixed. Additionally I
> Hi,
> i have a quick question.
> is it possible to use both XML dial plan and mod_perl together.
>
> examlpe:
>
> default.xml - used as default context
> mod_perl - used to generate the public context
>
> If it is possible how i have to set perl.conf.xml and especially
> xml-handler-binding
> Is there a way of displaying a console message not related to a log level?
> I’ve got the console only reporting errors now, but it would be nice to be
> able to display a message when a given condition exists. Yes, I could set
> it as an error level message, but I’d rather not do that.
What is
>
> to make sure that there is indeed a value and that it gets exported to
> the second b-leg try this:
>
oops, that set line should have been:
That "nolocal:" was extraneous from a lazy copy & paste
>
>
> see if my_var is populated on the new b-leg.
> -MC
>
The most elegant solution is the o
On Thu, Feb 26, 2009 at 1:18 PM, Tchavdar Paskov wrote:
> isn't this already done because of the info application called before ?
> Chav
>
to make sure that there is indeed a value and that it gets exported to
the second b-leg try this:
see if my_var is populated on the new b-leg.
-MC
___
>
Why are you using $0 here? Is that a typo?
-MC
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On Thu, Feb 26, 2009 at 10:51 AM, Rex_Alex wrote:
>
> Hi Mathieu,
>
> But other php scripts are working fine. Only when I am tring
> Single_Command.php with ESP.php, it's not working.
>
> Rex.
>
That may be true but the php-devel package is necessary for building
the ESL wrapper for php. The php-
On Wed, Feb 25, 2009 at 11:34 AM, Brian West wrote:
> If he's on 1.0.3 I don't think it has php in it..
Can't he do the whole bootstrap process?
svn up && ./bootstrap.sh && ./configure && make install
And then do Mathieu's suggestion?
-MC
___
Freeswi
> so far i am a happy camper with freeswitch. this is a different snack bracket
> than asterisk...
If you don't mind telling us, where did you hear about FS and what
made you decide to try it? Are you unhappy with Asterisk or are you
simply looking for something a bit different? Just curious.
Th
Klaus,
Can you update us on where you are with this?
-MC
On Sun, Feb 22, 2009 at 4:04 PM, Klaus Hochlehnert
wrote:
> Hi,
>
> I'm just playing around with FreeSWITCH and I have 2 questions about BLF
> (with SNOM phones):
>
> - When I played around with the sample dial plan I found out that BLF wo
> enter telephone number in webapp
> python script have fs to call number
> say "please enter the pin code from the website"
> validate dtmf code
> pass back to webapp: correct or not correct
>
> unfortunately just from reading the wiki i don't know how to do it in my
> python script.
>
> can you
On Tue, Feb 24, 2009 at 1:09 PM, Alexander de Greiff
wrote:
> hi all,
>
> i come from asterisk an i am new to freeswitch. after my with days with
> freeswitch i am very excited!
Welcome to FreeSWITCH!
>
> but trying to migrate our deployment i have three challenges. one of them is:
>
> i need t
> Maybe this question has been raised before, but if not: There's so
> much traffic in this mailing list that I was wondering if adding a
> web-based forum on the site was in the works?
We are upgrading the freeswitch.org site soon to drupal 6.9. We are
considering turning on the forum feature the
On Tue, Feb 24, 2009 at 9:24 AM, Ali Al-Rubaie wrote:
> Hi,
>
> The file directory.conf.xml had been mentioned in the documentation many
> times but there is not such file in the conf folder. Do you mean default.xml
> in directory folder?
>
> Thanks!
Can you tell me where you see that file name l
> Hello,
> if Cepstral 4.x is the way to go does anybody know where to get the demo
> version?
>
> BRs,
> Claudio
I think you'll have to contact Cepstral on this one. I've tried to
find older revisions on their site and I can't find any way to get any
voices prior to 5.1.
-MC
On Thu, Feb 19, 2009 at 5:04 AM, Rene Pankratz
wrote:
> Hello,
> when hanging up a call with portaudio automatically the next call that
> is incoming or held is accepted.
> Is it possible to configure PA that way, that after hanging up (doesn't
> matter whether caller or callee) no call is activat
On Thu, Feb 19, 2009 at 9:27 PM, Stephen Crosby wrote:
> I have a few scripts that use the javascript
> session.streamFile('somefile.wav', onDtmf); where onDtmf is a function
> that returns false to interrupt the streaming file. There is a short
> delay between the time when I press a key and the
On Thu, Feb 19, 2009 at 12:17 PM, Raymond Chandler
wrote:
> did you ./configure --enable-core-odbc-suport... those errors reek of
> that flag with no unixODBC-devel package installed
>
> -Ray
>
Anthony described this as a false positive on detecting ODBC. If you
are in Linux you can install the OD
On Thu, Feb 19, 2009 at 2:13 PM, jonathan augenstine
wrote:
> I have heard a rumor that Pika support was being developed for Freeswitch.
> Is that still going on? Can someone tell me if the rumor is true or not,
> and if so, what is the status of the development?
Well, the PIKA cards work with F
This is really advanced stuff. You're going to need to pay someone who
really understands DSP and programming. You might want to start with
consult...@freeswitch.org.
-MC
On Thu, Feb 19, 2009 at 4:58 AM, Frank @ Impact wrote:
> I am trying to detect if a caller is an automated greeting voice. A
On Wed, Feb 18, 2009 at 6:57 PM, Brian West wrote:
> Thats one I think Anthm will need to chime in on... maybe skypiax
> isn't sending the right indications to cause the core to trigger the
> ringback.
>
> /b
>
Out of curiosity, you might try this trick:
See also:
http://wiki.freeswitch.org/wiki/
> Everything is working perfectly, except the bridge to another number.
> Because of the nature of the beast the bridge needs to dial an external
> number (ie sofia/gateway/Mygateway/num) What I'm getting is:
>
> attempt to perform arithmetic on global 'sofia' (a nil value)
>
Can you pastebin you
On Wed, Feb 18, 2009 at 11:53 AM, Nik Middleton
wrote:
> I'm trying to build an emergency broadcasting solution.
>
> So I place a call, and have ivr in the lua script. But I also want to
> give them the option of speaking to someone.
>
> If they hit the option to speak to someone, while I can fir
On Wed, Feb 18, 2009 at 7:55 AM, Brian West wrote:
> Please go get an SVN client for windows... svn update vs downloading the
> tarball every day will save bandwidth. ;)
> /b
Use this for windows:
http://tortoisesvn.tigris.org/
-MC
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