d_number go into the action so you could leave it as $1, not
> 100 in my case?
>
> That way you could only have one of these plans work for all extensions.
>
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Lei Tang wrote:
> Hi All:
>I'm a newbie to FS. I'm using FS as a sbc and have about 2 user
> account . Does somebody can tell me how to make FS load use account
> information from a database such as mssql or mysql? Could you give me a
> sample configuration fil
ot;; bindings="dialplan"/>
> ...
>
> -- index.php --
>
>
> $_REQUEST['hostname'] = "$xhostname";
>
> ...
>
> - Original Message -
> *From:* freeswitch noob
> *To:* freeswitch-users@lists.freeswitch.org
> *Sent:* Friday, October 23
gt; > On Fri, Oct 23, 2009 at 3:34 PM, freeswitch noob
> >
> > wrote:
> > > I have mod xml_curl installed and I am getting the following passed to
> my
> > > script.
> > >
> > > [hostname] => myhost.local
> > > [section] => dialplan
dvance.
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y direction you guys could point me in would be great.
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documented on
the Zaptel Tutorial wiki page
(http://wiki.freeswitch.org/wiki/Zaptel_Tutorial). I'm using a Digium TDM400P,
Zaptel 1.4 revision 4630, and FreeSWITCH trunk revision 14842. If you want me
to try anything for you, I'm 'Deeewayne&#x
er");
+
switch_channel_state_t state_i = CS_DESTROY;
if (!switch_strlen_zero(state)) {
@@ -492,7 +504,9 @@
" read_rate VARCHAR(255),\n"
" write_codec VARCHAR(255),\n"
"
P
>
>
> Best regards.
>
> Dome C.
>
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Dear All,
How to set ${local_ip_v4} to 0.0.0.0 ? i want to set up
external profile for All IP
Best regards.
Dome C.
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thanks allot, this was my mistake.
/Tzury
> It tells you to edit conf/directory/default.xml not dialplan/
> default.xml and put
>
>
> as the dial-string.
>
> /b
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the code and looking at some of the simpler
modules. Most of FreeSWITCH is remarkably readable (just steer clear of
the XML parsing stuff :) ).
Andrew
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e format , and the IVR flow ??
*>>* ___
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*>>* Freeswitch-users at
lists.freeswitch.org<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
*>>* http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
*>
any timers on calls or specically on
recording calls. I'm using svn rev 13471 on debian, the calls originate from a
cisco 53xx media gateway and are using sip / g.711 ulaw.
Thanks.
Dan-
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I couldn't get the gateway to work with register=false. I took the hint
from Brian and scraped the gateway idea and put this in my dialplan.
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end the password for directory entry
> requests.
>
> try this:
>
>
>
> _______
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> UNS
1) How do you setup a gateway where the provider doesn't require a user name
and password? For freeswitch gateways username and password are mandatory.
2) Is there anyway to have xml_curl send the password for directory entry
requests.
3) What is the best way to monitor for failed sip reg
t does now which is presumably the first piece only
2: have the uninstall target back up your current binaries before removing them, and include some info about restoring them, for us idiots who neglect to do such things as
tar czf "freeswitch-bin-`date`.tgz" /usr/local/freeswitch/bi
rom: "Rupa Schomaker"
To: freeswitch-users@lists.freeswitch.org
Sent: Thursday, April 2, 2009 8:05:38 AM GMT -07:00 US/Canada Mountain
Subject: Re: [Freeswitch-users] mod_shout delay in trunk
2009/4/2 Anthony Minessale < anthony.miness...@gmail.com >
Its the buffering and s
Pablo Hernan Saro wrote:
> Hi list,
>
> Any experience building FS in Solaris using Sun Studio?
http://www.voiceworks.pl/cypromis/tag/opensolaris/
Chris
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no, I'm matching the incoming sip call via the destination number in my public
context and executing the javascript appliaction. This app directly answers the
call and records it until the user hangs up.
D-
- Original Message -
From: "Anthony Minessale"
To: f
That worked great!
I wanted to say just how awesome Freeswitch is, I have been doing voip related
development with SIP since 2000 and this is by far the most well designed piece
of voip software I have used or developed on. I currently have a homegrown sip
server built on the NIST sip stack
this worked very well.
Is this buffer configurable? If not, where is it being set?
Thanks
Dan-
- Original Message -
From: "Brian West"
To: freeswitch-users@lists.freeswitch.org
Sent: Friday, February 6, 2009 12:47:53 PM GMT -07:00 US/Canada Mountain
Subject: Re: [F
dial plan hits my record
statement I see mod_shout logging that it has connected to the icecast server,
in trunk it takes about 5 seconds to see the same log mesage. Below is my
current svn info
Path: .
URL: http://svn.freeswitch.org/svn/freeswitch/trunk
Repository Root: http://svn.freeswitch.org
Thanks, that's what I'm currently doing now.
- Original Message -
From: "Anthony Minessale"
To: freeswitch-users@lists.freeswitch.org
Sent: Wednesday, January 21, 2009 12:28:19 PM GMT -07:00 US/Canada Mountain
Subject: Re: [Freeswitch-users] Recording ULAW files
ster a new file type, like .ul? If so, where would be a good
starting point.
- Original Message -
From: "Anthony Minessale"
To: freeswitch-users@lists.freeswitch.org
Sent: Wednesday, January 21, 2009 10:56:45 AM GMT -07:00 US/Canada Mountain
Subject: Re: [Freeswitch-users]
ile.pcmu -t wav -r 8000 -c 1 -b -U
file.wav
it produces a file that shows:
/tmp/185065_f7bb8e0c-e641-11dd-800d-5ffe41c540dd.wav: RIFF (little-endian)
data, WAVE audio, ITU G.711 mu-law, mono 8000 Hz
if I try and save it within freeswitch using the wav extension, it trans-codes
it to a p
our phones.
I have just purchased one of these devices on eBay. Will it be able to ring all four of the dumb unpowered analog telephones in my house?
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I have just purchased one of these devices on eBay. Will it be able to ring all four of the dumb unpowered analog telephones in my house?
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?
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