quot;**" hitting
a single "*" would active it. Kind of a hack, but if no one comes up with a
more elegant way, I could provide a .patch file that did the necessary
changes.
I'd love an all-LUA method, or something that could use an existing
InputCallback routine, but this
t will create a channel. In my case I want to send/receive a message to/from the user freeswitch registered without a call being placed.Unless, during the registration, FS created a UUID for the open XMPP connection, in which case how would I find that UUID?-pete
Original Message -
appreciated.-pete
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ul to create a section of the FS Jira for larger projects like this as well as some of the bounty projects listed in the wiki, as a means of collaberation. That way we can aggregate requirements from many people.
-pete
Original Message Subject: Re: [Freeswitch-users] skill-b
o fund all/part of it. Other are just content with getting the job done. I offer my help in this area if the devs are interested in exploring this style of fund raising. -pete
Original Message
Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
From: Raimund Sach
I received a similar error trying to compile cepstral on a 64-bit OS. I don't think Cepstral supports 64-bit, as the SDK is 32-bit.-pete
Original Message
Subject: Re: [Freeswitch-users] Troubles build with mod_cepstral
From: Max Bridgewater
Date: Sun, August 30, 2009 11:
significant and cost a LOT more. I like the idea about the complete custom chassis. I hadn't considered that due to my thinking it would be expensive. Sounds like it's worth a look. As we consider creating an appliance offering, this may become more important.-pete
Original Me
ility, OpenSIPS was our choice. There's a very nice tutorial on their website on how to configure Load Balancing.-pete
Original Message
Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
From: "Pete Mueller"
Date: Sat, August 29, 2009 2:25 pm
s do not drop, but I will stress that I'm only running test cases at this time, I am not using real world traffic.Once I figure it all out, I'll report it here.-pete
Original Message
Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
From: Michael Collins
Date:
just transcribe the whole thing, or fix up what the computer spit out. If you have any insights on this, that would be great.-pete
Original Message
Subject: Re: [Freeswitch-users] VoiceMail transcription
From: David Knell
Date: Mon, August 10, 2009 11:51 am
To: f
s for any help-pete
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I believe you need to set LUA_PATH, here's more information:http://www.lua.org/pil/8.1.html-pete
Original Message
Subject: [Freeswitch-users] Lua on Windows and additional modules
From: Vladimir Rodionov
Date: Thu, August 06, 2009 5:55 pm
To: freeswitch-
Yes, you can use the stream global object. example: local api = freeswitch.API(); local reply = api:execute("originate", someRoute); if (reply) then stream:write("RESULT: " .. reply .. "\n"); else stream:write("ERROR") end
Original Message
Subject: [Freeswi
en performs a lookup for the extension to route to.If you would like a copy of my switchboard script I can provide it to you in a PM.-pete
Original Message
Subject: Re: [Freeswitch-users] Multiple DIDs per SIP trunk (how to
configure?)
From: Vladimir Rodionov
Date: Wed, August 05,
before beginning my logic, I can avoid that.
Original Message
Subject: Re: [Freeswitch-users] Confusing handling of incoming calls
From: Rupa Schomaker
Date: Wed, July 22, 2009 6:04 am
To: freeswitch-users@lists.freeswitch.org
On Wed, Jul 22, 2009 at 5:11 AM, Pete Mueller &l
dialplan would be a mess. AFTER I know what gateway the call arrived on, I have a database for each gateway that helps me process from there.2) Yes, separate profiles would work, but does sound gross. I'm going to swap my ports around and see if that clears things up... -pete
Ori
I have two different gateways setup on my server. One with FlowRoute (which uses SIP REGISTER) and one with bandwidth.com (which does not). I can send and receive from both gateways, but the dialplan processing seems to be confused. Calls from bandwidth do not respect the "Extension" param in th
I did a fair amount of research into GSM gateways about 8 months ago. I should first ask what are you looking to do with the gateway?-pete
Original Message
Subject: [Freeswitch-users] Which GSM gateway to buy?
From: Diego Viola
Date: Tue, June 16, 2009 2:39 pm
To: freeswitch
: freeswitch-users@lists.freeswitch.org
Thanks guys for a detailed reply specially pete. On Tue, May 19, 2009 at 7:31 PM, Peter P GMX <prometheus...@gmx.net> wrote: Thanks for this overwiev.One question: How does this compare to Cepstral TTS?Best regards Peterp...@privateconnect.com schrieb:
s, and that is a completely subjective decision.-pete
Original Message
Subject: [Freeswitch-users] text to speech IVRs and MOH
From: Saeed Ahmad
Date: Tue, May 19, 2009 12:40 am
To: freeswitch-users@lists.freeswitch.org
Hi all, Could you guys recommend me any online text to s
handy benefit for LUA, you can configure LUA scripts to run at switch startup to perform tasks (like a cron system) within the switch core. -pete
Original Message
Subject: Re: [Freeswitch-users] help with mod_conference stability
From: Brian West
Date: Thu, May 14, 2009 10:31
I looked at bind_meta_app, unfortunately, it requires a two character sequence. My requirement is to break the call on a single key press (either the "*" key, or ANY key press would be acceptable)-pete
Original Message
Subject: Re: [Freeswitch-users] Detecting DTM
ources.
However, if the actual resource usage for conference vs. bridge isn't really
noticeable, then maybe conference is the way to go.
I thought I figured out how to do this one day, but now I can't remember how
J
-pete
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d for hours at a time, and then everything goes back to normal. Thanks for your help-pete--- BEGIN SCRIPT ---function cbPIN(ses, type, data, arg) if (type == "dtmf") then freeswitch.consoleLog("info", "DIGIT: " .. data.digit) local d = ses:getVariable(
time, and then everything goes back to normal. Thanks for your help-pete--- BEGIN SCRIPT ---function cbPIN(ses, type, data, arg) if (type == "dtmf") then freeswitch.consoleLog("info", "DIGIT: " .. data.digit) local d = ses:getVariable("private_digits&quo
session:setInputCallback() to receive tones, did not
test with playAndGetDigits()
Thanks for any help.
-pete
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Hi,
If I have different audio tracks that I would like to loop through, does it
mean I need different folder for each? Moreover, if I loop through
different folders, does it mean freeswitch would start multiple "player" and
consume resources ineffectively as the audio is probably played only for
all
that reaches the other boxes? I don't totally understand how to "distribute
the PARKed calls to satellite machines"...
Thanks for your advice.
Regards,
Pete
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ow to best approach this problem?
Thanks in advance for your help.
Best Regards,
Pete
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, please kindly donate the bounty to the
Freeswitch project.
Thanks,
Pete
On Wed, May 28, 2008 at 11:26 AM, Klaus Teller <[EMAIL PROTECTED]> wrote:
> Hi Folks,
>
> I have 30$ bounty for the following task. I want a FS profile to support
> DIDWW. I intend to use Les.net for outbound c
Then, I filtered on all the events, but I can't find anything that has
the text "main", "menu", "custom" or "message".
I am wondering if my code has bug or I was not looking at the right
place in my event log.
Could someone
BUG from sofia/default/
[EMAIL PROTECTED]:5060
Can someone please kindly tell me what the problem could be and how can I
fix it?
Thanks alot for all you kind input.
Best Regards,
Pete
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"
command, but it giving ERR. So, which uuid should I be using?
Thank you in advance for your kind inputs.
Regards,
Pete
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#x27;);
$xmlw -> writeAttribute('id', $_REQUEST['sip_auth_username']);
So, it seems like in voice mail main, those data are passed in as "".
That is why I want to use the "auth ..." option to check vm, but I can't get
"auth" to work.
Any
:
Since I am using the user 1005 to dial in, so userid and password should
work, otherwise, I won't be able to login to dial the number.
Any inputs will be greatly appreciated.
Regards,
Pete
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fetch the first user from the queue (using fifo out ) ?
How is it possible for the agent's phone to ring automatically when a user
is added to the queue?
If someone can show me with an example, it will be greatly appreciated.
Regards,
not good.
Is there any good way to search for an element within a resultset?
Thank you in advance for your input.
Regards,
Pete
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play the wav file
In a highly-loaded environment, which option would be preferable considering
speed and efficiency?
Is there other better option?
Any input will be greatly appreciated.
Thanks,
Pete
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working under a load balancing
situation?
Thanks in advance for your input.
Thanks,
Pete
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find the place to
change it since it is a standard feature.
The other question is about TTS engine. I just want to know what is the
best way to play media coming out of my own TTS engine?
Thank you very much for all your help in answering my questions.
Regards,
Pete
On Sun, Apr 20, 2008 at
Hi,
I two a few questions about hangup and sounds:
1.
In my dialplan, I have
After it play the call-back-later marco, it then goes on to play the "The
extension you are dialing is unavailable..." even I have specified the
HANGUP application.
() Error loading
ODBC
2008-04-20 03:15:09 [ERR] switch.js:7 mod_spidermonkey() ReferenceError:
ODBC is not defined
Thank you very much for your kind help and input.
Regards,
Pete
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he etpan? Will the voicemail.js work without it?
Is etpan needed and how to get it to work? I can't find the module etpan.
Thanks alot.
Pete
On Fri, Apr 18, 2008 at 10:05 PM, Brian West <[EMAIL PROTECTED]> wrote:
> This is a huge clue.. check permissions.
> /b
>
> On Ap
008-04-19 03:40:20 [DEBUG] inc_logger.js:6 console_log()
>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> session.streamFile rtn=[undefined] >>>>>>>>>
undefined
What did I do wrong?
Thanks in ad
t; account dir = /var/spool/freeswitch/1
>>>>>>>>> undefined
2008-04-19 03:26:28 [ERR] answermachine.js:52 mod_spidermonkey() Error:
File operation mkdir failed
Thank you very much for your inputs.
Regards,
Pete
_
:29:38 [DEBUG] switch_core_state_machine.c:65
switch_core_standard_on_ring() Standard RING sofia/default/
[EMAIL PROTECTED]:5060
2008-04-19 01:29:38 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing
1001->[EMAIL PROTECTED]
Any inputs will be greatly appreciated.
Regards,
P
Hi,
>From the debug file created by fs , I can see the response as follows:
But it still does not authenticate my user. Why? I have checked the format
which is same as the one in the wiki doc.
Thanks,
P
Hi,
I have a questions about multi-language voice support in FS. How to I add a
new module such as mod_fr.so? I can make the necessary wav files, but how
to create the modules?
Thank you very much for your inputs.
Regards
Pete
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am suspecting that the info pass is not
sufficient for generating the directory xml. Is there anyway to see the
HTTP request and response from the console? I turned on curl_xml debug_on
but still it does not show.
Any inputs will be greatly appreciated.
Thanks,
Pete
On Fri, Apr 18, 2008 at 2:28 PM
62d8d5f9cd7e&sip_auth_method=REGISTER&key=id&user=1007&domain=192.168.1.104&ip=
192.168.1.103]
I think this may just be a very minor error in my setup.
Can someone please tell me what I did wrong?
Thank you very much in advance for your help.
Pete
__
) calling sip client ( 1004) results in 1001->
1004/public
wherease sip client ( 1004) calling xLite ( 1001) results in
1004->1001/default
Don't know why.
The setting was default.
Thanks for your help in taking a look.
Pete
On Fri, Apr 18, 2008 at 1:51 PM, Brian West <[EMAIL PROTECT
read() Close Channel sofia/default/
[EMAIL PROTECTED]:5061 [CS_HANGUP]
2008-04-18 08:53:16 [DEBUG] sofia.c:219 sofia_event_callback() event
[nua_r_refer] status [408][Request Timeout] session: n/a
Thanks for your help.
Regards,
Pete
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Hi,
I comment the line at default.xml
But, it still does not work.
2008-04-18 07:38:32 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing
1002->[EMAIL PROTECTED]
2008-04-18 07:38:32 [INFO] switch_ivr_async.c:1357
switch_ivr_bind_dtmf_meta_session() Bound: 1 execute_extension::
Hi,
I created another callwithus.xml under outbound and it is working now.
Thanks alot.
Pete
On Thu, Apr 17, 2008 at 11:27 PM, UV <[EMAIL PROTECTED]> wrote:
> Try:
>
> Sofia status gateway callwithus
>
>
>
> (maybe someone should add this to the wiki page
> h
Hi Brian,
What is the difference between $$ and $ in FS?
Thanks,
Pete
On Thu, Apr 17, 2008 at 11:11 PM, Brian West <[EMAIL PROTECTED]> wrote:
> You could also use user/[EMAIL PROTECTED]
>
> in addition you could do sofia/$${domain}/${dialed_ext}
>
> /b
>
>
>
>
Hi,
When I do reloadxml, I keep on getting this error:
[EMAIL PROTECTED]> reloadxml
Error including /usr/local/freeswitch/conf/dialplan/extensions/*.xml
API CALL [reloadxml()] output:
Is this another bug in the sample files? If so , how do i fix it?
Thanks,
P
lease help me so I get get started with FS.
Thanks,
Pete
On Thu, Apr 17, 2008 at 10:48 PM, UV <[EMAIL PROTECTED]> wrote:
> Try this instead:
>
>
>
>
> --
>
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *P
Hi,
I am still not yet able to get one sip phone to call the other due to the
problem I posted. So, I used the working sip client to try to call an
outside number
by setting up a gateway.
I tried to set up fastswitch to route call to my voip provider, I am getting
channel error but can't know wh
I can debug the config easier?
Thanks alot for all your inputs and help.
Regards,
Pete
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Hi,
I made a silly mistake. The problem was solved. Please ignore.
Thanks.
Regards,
Pete
On Thu, Apr 17, 2008 at 12:38 PM, Pete Kay <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I followed the online installation but when I executed, make sure or
> make moh, I am getting th
oblem?
My installation procedure is exactly same as the freeswitch wiki.
Thanks,
Pete
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much in advance for your inputs.
Regards,
Pete
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-time does?
Thanks alot for your inputs.
Regards,
Pete
On Thu, Apr 17, 2008 at 12:48 AM, Brian West <[EMAIL PROTECTED]> wrote:
>
> On Apr 16, 2008, at 11:44 AM, Pete Kay wrote:
>
> > Hi,
> >
> > I have studied the Freeswitch doc and can't find any info r
the above three options, which one is more preferable in terms
of scalability? If Freeswitch is used, is Operser still needed?
Thanks alot in advance for your input.
Thanks,
Pete
On Wed, Apr 16, 2008 at 11:29 PM, Daniel Hefti <[EMAIL PROTECTED]> wrote:
> It's ok, I had an idea
k AMI-equivalent feature in Freeswitch? What
is that called?
Thanks for your input.
Pete
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inputs on this regards?
Thanks,
Pete
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http
Freeswitch better than Asterisk in terms of functionality,
ease-of-maintain, and ease-of-use?
Thanks alot in advance for your inputs.
Pete
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switch, should I be using it to replace Openser or part of the
routing that is done by Asterisk?
Thanks alot for your inputs.
Pete
On Fri, Apr 11, 2008 at 6:56 AM, David Knell <[EMAIL PROTECTED]> wrote:
> Hi Pete,
>
> Tell us a bit more about your application, and we'll be b
share
with me?
Thank you very much for your input.
Best Regards,
Pete
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