You could use mod_limit for this...
From: Scott Ellis
Reply-To:
Date: Thu, 22 Jan 2009 17:14:15 +1100
To:
Subject: [Freeswitch-users] Can I restrict a gateway to 1 call at a time?
I need to restrict the outbound calls to an SPA3000. Is there a way of
marking a gateway to only allow one call
I need to restrict the outbound calls to an SPA3000. Is there a way of
marking a gateway to only allow one call at a time?
Or, as the SPA3000 registers with FS, allowing the extension to only
allow one outbound call. (Problem here is I have not worked out how to
send a call to the FXS using th
I just used mono 2.2 src tarbal .. configured with --prefix=/usr
--sysconfdir=/etc --localstatedir=/var
And Freeswitch just configured as --prefix=/usr/local/freeswitch
Nothing special.
I did run ldconfig after editing files on the same shell as I used to start
freeswitch
I am manually loading mo
You have to make sure you do it in the same shell before you start
FreeSWITCH or it won't work.
How did you install mono on 5.2 I'll try it out also.
/b
On Jan 21, 2009, at 8:56 PM, Adam Long wrote:
> Yah I did try that ... doesn't seem to work.
> Which seems odd I had thought the same thing
Yah I did try that ... doesn't seem to work.
Which seems odd I had thought the same thing.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Wednesday, January 21, 2009 9:16 PM
To: fr
Bet you could add /usr/local/freeswitch/mod to /etc/ld.so.conf and run
ldconfig and accomplish the same thing ;)
/b
On Jan 21, 2009, at 8:07 PM, Adam Long wrote:
> Thanks Michael!
>
> FreeSWITCH.Managed.dll loads successfully without tinkering. Tks
> for the patch!
> Of course after adding
Thanks Michael!
FreeSWITCH.Managed.dll loads successfully without tinkering. Tks for the
patch!
Of course after adding the LD lib path.
Thank you very much!
Also saw you updated the BUG on Jira.. Thanks again great work!
Only other thing to note is "languages/mod_managed" still need
Most if not all of this functionality is done with FreeSWITCH without
any need for python in the first place.
For example you don't use python for voicemail, conferences, tone
generation but maybe IVR... which is what any of the languages are for.
/b
On Jan 21, 2009, at 7:00 PM, Stephen Cros
I'd like to see an example of recording voicemail and a good
conference application, tone generation, just a good IVR example would
be nice too.
--Stephen
On Wed, Jan 21, 2009 at 4:42 PM, Brian Deacon wrote:
> Well, rescuing someone from the evils of javascript cleans so many years
> off my time
Well, rescuing someone from the evils of javascript cleans so many years
off my time in purgatory that I couldn't possibly pass up the chance. :)
But I'm still probably another week or two from qualifying as N00b. I
imagine I could probably handle transcoding javascript examples to
python if it's
See, my powers of stupidity are legendary. :)
I updated the one-liner code sample on the mod_python page (near the
part that points to python_example.py) to include the def fsapi line.
Works like a champ now! Thanks!
Brian
On Wed, 2009-01-21 at 16:55 -0600, Anthony Minessale wrote:
> add
>
>
Yes and yes, I will get the full details to you next week, snowed under
a bit at the moment.
Scott
Anthony Minessale wrote:
did you answer the call in your dialplan?
do you have a full debug log of a call with that parameter enabled on
the analog span in question?
On Thu, Jan 15, 2009
I would love to see the documentation for python flourish. I would
have chosen python for my recent development if I could have quickly
figured out how to use it. I chose the javascript route instead
because of all the examples on the wiki.
--Stephen
On Wed, Jan 21, 2009 at 3:33 PM, Michael Colli
On Wed, Jan 21, 2009 at 2:40 PM, Brian Deacon wrote:
> Greetings,
>
> Couldn't find anything on the wiki or in the mail archives. (Let me
> know where you think a good home for this info might be on the wiki and
> I'd be more than happy to write something up in there.)
Thanks, you're hired! :)
We
add
def fsapi(session, stream, env, args):
stream.write("baz")
see:
http://svn.freeswitch.org/svn/freeswitch/trunk/src/mod/languages/mod_python/python_example.py
could make a good addition to the wiki
On Wed, Jan 21, 2009 at 4:40 PM, Brian Deacon wrote:
> Greetings,
>
> Couldn't fin
Greetings,
Couldn't find anything on the wiki or in the mail archives. (Let me
know where you think a good home for this info might be on the wiki and
I'd be more than happy to write something up in there.)
I'm guessing I haven't done everything necessary to enable python on my
machine. I have
Anthm,
I will add the substance of this to the wiki.
-MC
Sent from my iPhone
On Jan 21, 2009, at 2:09 PM, Anthony Minessale > wrote:
lua:
local event = freeswitch.Event("custom");
js:
e = new Event("custom", "message");
in js you specify a subclass which means you would need to subscrib
lua:
local event = freeswitch.Event("custom");
js:
e = new Event("custom", "message");
in js you specify a subclass which means you would need to subscribe to it
events plain custom message
really just
events plain custom is not a good idea because all the real events have a
subclass
once the
RESOLVED!
Stephen, when using event sub-classes you need to specify the subclass
when subscribing to listen to the events. I added a small entry to the
wiki page that I hope makes it clear:
http://wiki.freeswitch.org/wiki/Javascript_Event#Subscribing_To_Custom_Events
Let me know if you have any mo
Sure, I'll give it a try when I get home.
On Wed, Jan 21, 2009 at 1:43 PM, Michael Collins wrote:
> Stephen,
>
> I've been able to duplicate this behavior on my Mac with r11333. It
> seems to work with Lua but not with Javascript. I am going to discuss
> it with the devs and possibly open a jira
Stephen,
I've been able to duplicate this behavior on my Mac with r11333. It
seems to work with Lua but not with Javascript. I am going to discuss
it with the devs and possibly open a jira issue. In the meantime would
you be willing to try it with Lua, even just for testing? This worked
for me:
-
mod_sndfile already registered .ul and .al ;)
/b
On Jan 21, 2009, at 1:05 PM, freeswitch-us...@digitaldan.com wrote:
Thanks for the quick reply.
I'm new to this project so I'm not familiar with the inner workings
just yet but at looking at mod_native_file.c it seems this is a thin
wrappe
On Wed, Jan 21, 2009 at 12:21 PM, Dave wrote:
> Just like to know whether there is any kind of document available
> for FS integration at its C level, not at its dialplan level.
> The reason I ask this is that some of the local process simply
> impossible to be integrated with the Dialplan, but
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
The default audio f
Just like to know whether there is any kind of document available
for FS integration at its C level, not at its dialplan level.
The reason I ask this is that some of the local process simply
impossible to be integrated with the Dialplan, but should not be that
hard if the C level API is availab
did you answer the call in your dialplan?
do you have a full debug log of a call with that parameter enabled on the
analog span in question?
On Thu, Jan 15, 2009 at 4:17 AM, Scott Ellis wrote:
> After poking around in the code, it looks like if I set name="enable-callerid" value="false"/> in op
thanks we'll have a look. Also, please use pastebin.freeswitch.org in
the future because it makes it easier for us to find things. :)
-MC
On Wed, Jan 21, 2009 at 8:53 AM, Stephen Crosby wrote:
> Today I was able to see the event on the listener by subscribing to
> all events. But I'd like to only
Hi Jonathan,
Mod_vmd (voicemail detection) should do the trick.
Just search the wiki for mod_vmd, there are a number of ways of using it.
Sent from my BlackBerry device on the Rogers Wireless Network
-Original Message-
From: jonathan augenstine
Date: Wed, 21 Jan 2009 06:53:41
To:
S
The default audio framework operates on raw audio.
The mod_native_file triggers a special flag that tells the higher level
api's for recording not
to transcode the audio first.
It would probably be easier for you to use the api_hangup_hook variable to
trigger a sox command
to wrap the files in a w
Thanks for the quick reply.
I'm new to this project so I'm not familiar with the inner workings just yet
but at looking at mod_native_file.c it seems this is a thin wrapper around the
switch's own file input and output routines? Would it be best to change this
class or register a new file type
no, there is currently no way to do that.
It would be feasible to add an option to mod_native_file to write wav
headers around the raw data but it has not been attempted.
On Wed, Jan 21, 2009 at 10:48 AM, wrote:
> Hi, I'm recording files using the pcmu extension in order to save them in
> the
a normal call should flow like that, with the possible exception of
the ack handling, we don't wait for the a leg ack before we ack the b
leg and the same for the 200ok to bye going the other way.
Mike
On Jan 21, 2009, at 11:44 AM, Adam Long wrote:
Something like the following perhaps???
Hi, I'm recording files using the pcmu extension in order to save them in the
g.711 ulaw format, which is what everything in my network uses. It appears that
the recorded file is just raw data without a header. Is there any way to save
this as a wav type with a header (keeping the ulaw format)?
Today I was able to see the event on the listener by subscribing to
all events. But I'd like to only subscribe to a subset if possible. I
thought that it would pop up when subscribing to CUSTOM events. I've
put it all together very neatly here:
http://pastebin.com/m6f8f7b43
--Stephen
On Wed, Jan
You can already do this ... its how a phone call already works.
CALL A -> FS -> CALL B
Call A will answer when Call B is picked up passing the answer over to
Call A.
/b
On Jan 21, 2009, at 10:14 AM, Kareem Hamdy wrote:
> Hello everyone:
>
>I think what Anthony wants is (please excu
eeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.org
pstn:213-799-1400
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If you can get it to work in openzap, it will be easy enough for me to
port when we do the windows driver integration for openzap.
Mike
On Jan 21, 2009, at 10:01 AM, Helmut Kuper wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hello,
>
> It's a standalone program to profe concept.
Been running all day without crashing. Will see on the long term.
-- Laurent FABRE
Directeur général
10, rue d'Aumale
75009 Paris
Tel: +33.(0)1.42.81.28.20
Mob: +33.(0)6.75.75.02.96
Fax: +33.(0)1.70.24.74.61
laurent.fa...@kirranet.com
-Message d'origine-
De : freeswitch-users-boun...@lis
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
It's a standalone program to profe concept. So I think I will get it
working this week as standalone.
Next week I plan to start to integrate the code into openzap/ozmod_isdn.c
Biggest problem currently (in my head) is the file handling of the
On Wed, Jan 21, 2009 at 1:49 AM, Ognjen Seslija wrote:
> When call comes in from Openzap, tone_detect app does pre_answer of a call
> cause it's need media to start detecting tones in the first place. This
> behaviour is something that I see on calls inside my telco when calling from
> analogue li
On Wed, Jan 21, 2009 at 1:36 AM, Scott Ellis wrote:
> I had a similar problem, you can use
> (I added an "au"
> ring definition to my vars.xml file)
>
> To get what you want.
>
> I also had a problem that you get two rings, then an answer then to the
> system generated ring tone, which was confu
I have an application that requires answering machine detection. I have not
been able to locate any documentation indicating that there is explicit
support for answering machine detection. I have received recommendations on
call flows that would include DTMF entry by the called party to detect by
Okay, try the changes I note below
-MC
On Wed, Jan 21, 2009 at 6:33 AM, Krzysztof Zimnicki wrote:
> conf/openzap.conf
>
> [span zt]
[span zt PRI_1]
> name => OpenZAP
> number => 1
> trunk_type => E1
> b-channel => 1-15
> d-channel => 16
> b-channel => 17-31
> [span zt]
[span zt PRI_2]
> name =>
conf/openzap.conf
[span zt]
name => OpenZAP
number => 1
trunk_type => E1
b-channel => 1-15
d-channel => 16
b-channel => 17-31
[span zt]
name => OpenZAP
number => 2
trunk_type => E1
b-channel => 32-46
d-channel => 47
b-channel => 48-62
On Wed, Jan 21, 2009 at 2:48 PM, Michael Collins wrote:
>
can you create a pastebin with the two scripts in question? We'll take
a look and see if we can figure out what's going on.
Thanks,
MC
On Tue, Jan 20, 2009 at 11:04 PM, Stephen Crosby wrote:
> I noticed the wiki has an example of sending a custom event from
> javascript: http://wiki.freeswitch.or
You can't.
Why would you need that? Are you trying to forward inbound calls from the
pstn to an ivr without answering them?
That could get you in trouble FYI.
On Wed, Jan 21, 2009 at 7:40 AM, shehzad p wrote:
>
> Hi all,
>
> When I dial a number from Originator Gateway, It will route to Frees
Do you have it integrated into openzap or just standalone?
We can give you commit access to openzap if you want to maintain the patch
in tree if you'd like.
On Wed, Jan 21, 2009 at 5:58 AM, Helmut Kuper wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hello,
>
> just an update about
On Wed, Jan 21, 2009 at 4:38 AM, Michael Giagnocavo wrote:
> You're right, there should be a full installer system that'll ask what
> account you want to use, check permissions, etc.
> -Michael
Has somone opened a feature request issue on jira yet? If not, I
highly recommend that you do so
-MC
On Wed, Jan 21, 2009 at 3:58 AM, Helmut Kuper wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hello,
>
> just an update about my progress in this.
>
> Currently I have working C code which creates a pcap file containing all
> needed protocoll headers in front of the Q931 dump. I use l
On Wed, Jan 21, 2009 at 1:23 AM, Tomás wrote:
> Thanks Michael, I think that's very useful information for newbies like
> me...
>
> On the other hand, I'm looking forward to the OpenZap section because I'm
> having some problems with my card, and maybe in the new section I could find
> useful info
can you post your openzap.conf file?
-MC
On Wed, Jan 21, 2009 at 12:55 AM, Krzysztof Zimnicki wrote:
>>Can you join irc later today? I will be on as mercutioviz. I would
>>like to discuss this more.
>>
>>-MC
>
>>Sent from my iPhone
>
> Sorry, i can't join to irc. Can you put your questions here?
Hi all,
When I dial a number from Originator Gateway, It will route to Freeswitch
Server and then FS will bridge the call to Terminator Gateway as below.
Terminator Answer the call (and runs playback, and look for DTMF).
|Originator Gateway|---> |FreeSwitch |-->
|Term
You're right, there should be a full installer system that'll ask what account
you want to use, check permissions, etc.
-Michael
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Ron Avriel
Sent: Tue
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
just an update about my progress in this.
Currently I have working C code which creates a pcap file containing all
needed protocoll headers in front of the Q931 dump. I use libpcap to
create the pcap file. The protocol addresses of each protoc
When call comes in from Openzap, tone_detect app does pre_answer of a call
cause it's need media to start detecting tones in the first place. This
behaviour is something that I see on calls inside my telco when calling from
analogue lines. I don't think this is a big of deal because ringback
provid
I had a similar problem, you can use
(I added
an "au" ring definition to my vars.xml file)
To get what you want.
I also had a problem that you get two rings, then an answer then to the
system generated ring tone, which was confusing some of our (not to
bright) callers.
As we don't use ca
Thanks Michael, I think that's very useful information for newbies like
me...
On the other hand, I'm looking forward to the OpenZap section because I'm
having some problems with my card, and maybe in the new section I could find
useful information... And If you need a tester with a X101P card, you
Scott, I imagined that it could be an OpenZap problem, but I didn't find an
OpenZap mailing list, so I sent the email to FS list. Do you know where can
I find more information about OpenZap hardware support and developement
status (I have special interest in Loop Start)??
Anthony and Ognjen, I've
>Can you join irc later today? I will be on as mercutioviz. I would
>like to discuss this more.
>
>-MC
>Sent from my iPhone
Sorry, i can't join to irc. Can you put your questions here? I'll try to
answer.
Our CallCenter have strange situation, because now is working on Asterisk and
we can o
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