Re: [Freeswitch-users] mod_conference voice problems when two parties speaking

[Dan Le]

No, from my understanding that's not how it works. Waste just means it'll
always send RTP packets, doesn't mean it will contain audio... so if you
have audio that's under your energy threshold, you still won't hear it.

Dan

On Wed, Dec 23, 2009 at 12:45 AM, Marc Orenberg  wrote:

> Thanks Rob, thanks Jason.
> I'm going to try this first thing tomorrow.
> The "energy-level" paramter is described in the file as, "Energy level
> required for audio to be sent to the other users", so one would think that
> this would have no effect if member-flags is set to "waste", right?
>
> --
> *From:* Jason White 
> *To:* freeswitch-users@lists.freeswitch.org
> *Sent:* Wed, December 23, 2009 12:29:58 AM
> *Subject:* Re: [Freeswitch-users] mod_conference voice problems when two
> parties speaking
>
> Marc Orenberg  wrote:
> > Is there something in mod_conference which would try to filter out other
> > voices when one voice is speaking?
>
> Try reducing the energy level parameter in case this is the issue. It's
> 7/8/9
> on the key pad during the call, or via the conference command, or the
> settings
> in conference.conf.xml.
>
>
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Re: [Freeswitch-users] PlayAndGetDigits multiple WAV files

[Dan Le]

Or using mod file string: http://wiki.freeswitch.org/wiki/Mod_file_string

Dan

On Tue, Dec 15, 2009 at 5:05 PM, Michael Jerris  wrote:

> You can do that with phrase macros.
>
> Mike
>
> On Dec 15, 2009, at 2:56 PM, Malay Thakershi wrote:
>
> Hello, I create one WAV file that has:
>
> Question + Option 1 + Option 2 + Option 3 + …
>
> I noticed towards end of the file Cepstral Allison starts chopping and
> speeding up.
>
> So my question text that gets converted to WAV file using swift EXE looks
> like:
>
> Which is the biggest mammal on land?
> Select one of the following choices. strength='weak'/>Or press star to skip the question
> 1  Parrot
>  2  Elephant
>  3  T-Rex
>  4  Blue Whale
> 
>
> And my csharp code looks like:
> pStrRetID = mObjMainSession.PlayAndGetDigits(1, 1, 3,
> 5000, "*#",
> 
> @"C:\FreeSWITCH\sounds\en\us\chAsmt\Student_1222\Q1.WAV
> ",
>
> @"C:\FreeSWITCH\sounds\en\us\chAsmt\static\error\invalid_choice.wav",
> "^\\d", "");
>
>
> What happens is, the voice just starts chopping and speeding up between
> options. Even though I am not able to say that it only does that towards the
> end, I think so.
>
> I thought, if I break each file into individual WAV instead of 1 big WAV,
> it may help?
>
> Is there a way to play multiple (separate) WAV files in PlayAndGetDigits
> function?
>
>
>
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Re: [Freeswitch-users] Minimum audio length for uuid_record

[Dan Le]

Thanks, I think I found the thread you were referring to
("[Freeswitch-users] session record does not for very short calls"), which
doesn't seem to be a solution for my situation. However, I did find that
using session:recordFile() didn't delete the file if it was really short.
And following that thread lead me to an interesting channel variable that
could be useful to us, record_ms, but having trouble getting it to reflect
the audio length. I can see the variable when printing data from the info
application, but it's always 0. My snippet of code is very simple:
if session:ready() then
   session:recordFile("C:/Temp/recording.wav", 30, 600, 6);

   local record_length = session:getVariable("record_ms");
   freeswitch.consoleLog("INFO", "Recorded a " .. record_length .. " ms
file.\n");
end

Another side question, the silence secs parameter (in this example, 6), is
that 6s silence hits during the entire recording session or 6s of
consecutive silence? From a few tests, it seems to be the former, but just
wanted to verify, something that would make a good addition to the wiki.

Dan


On Wed, Sep 30, 2009 at 11:31 PM, Jason White  wrote:

> Dan Le  wrote:
> > We're running into a problem with the minimum file size when recording
> using
> > uuid_record. It seems if the audio is too short it deletes the audio
> file.
> > Is there a way to override that?
>
> Yes. It was discussed on the list recently. I suggest searching the list
> archives. Someone may have documented it on the wiki by now also.
>
>
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[Freeswitch-users] Minimum audio length for uuid_record

[Dan Le]

We're running into a problem with the minimum file size when recording using
uuid_record. It seems if the audio is too short it deletes the audio file.
Is there a way to override that?
We're running the FS 1.0.4 tag, on a windows xp 64bit machine. We start
uuid_record from a lua script, recording at 8khz, if the recording is
shorter than about 5s, the file gets blown away, but we may want to keep
recordings that are 3-4s long.

Thanks,

Dan
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Re: [Freeswitch-users] Best practice for inbound calls with scripting

[Dan Le]

One common recommendation is to use lua over js, since it's lighter-weight,
using less resources.
Dan

On Wed, Aug 5, 2009 at 1:31 PM, Greg Thoen wrote:

> Hi. I am setting up a large inbound only system with multiple DIDs coming
> in; each call is processed fairly intensively with db lookups, wav files
> played, pocketsphinx is used, wav files recorded, etc.
> Before I get too far down one path, I was wondering if anyone had any
> insight into the best, most scaleable way to do this out of the several
> methods I can do:
> 1 Call comes in
> dialplan calls specific javascript based on DID
> javascript uses ODBC to pull info from local mysql db
> call is handled in javascript
>
> 2 Call comes in
> dialplan calls specific javascript based on DID
> javascript uses CURL to get info from local mysql db
> call is handled in javascript
>
> 3 Call comes in
> php socket is listening for call
> php script runs, pulling info from mysql db
> call is handled in php using esl.php
>
> 4 Call comes in
> dialplan calls specific lua script  based on DID
> lua uses luasql.mysql to get info from local mysql db
> call is handled in lua using lua api
>
> 5 Call comes in
> xml_curl is used for dynamic dialplan
> js called, continues like #1
>
> I know that they will all do
> essentially the same thing. But once I go down the path, I don't want to find 
> out that the way I chose chokes with 50 simultaneous inbound calls on 
> different DIDs. Any comments would be appreciated. Thanks.
>
> --
>
> Greg
>
>
>
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Re: [Freeswitch-users] How to call a non-user extension from "originate" [SOLVED]

[Dan Le]

Technically, loopback is not meant to specifically allow you to call
non-user extensions, it simply allows you to hit the dialplan.
For example,

originate sofia/gw/gwname/fakecall 1000

(where gwname is the gateway routing your calls out, and 'fakecall' the
non-user extension)

will also allow you to dial to non-user extensions. This information would
be on any of the wiki pages detailing the originate command.

Dan


On Mon, Aug 3, 2009 at 3:13 PM, Raffaele P. Guidi <
raffaele.p.gu...@gmail.com> wrote:

> Yeah, it's there but, as you can see it from the google queries below it's
> not easy to find that page - unless you know what exactly you are looking
> for and search for "loopback". A simple example (i.e. in Sofia Syntax) would
> fill the gap (I'll be happy to do it ASAP)
>
>
> http://www.google.com/search?hl=en&safe=off&q=freeswitch+call+url&aq=f&oq=&aqi=
>
>
> http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+cli+&aq=f&oq=&aqi=
> 
> http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+esl&aq=f&oq=&aqi=
>
> Also, this email now shows up first in googling for "non user extension
> freeswitch" (which were the keywords I was looking for). I think this has
> been useful! ;)
>
> Many thanks and regards,
>Raffaele
>
> On Mon, Aug 3, 2009 at 14:59, Michael Collins  wrote:
>
>>
>>
>> On Mon, Aug 3, 2009 at 4:43 AM, Raffaele P. Guidi <
>> raffaele.p.gu...@gmail.com> wrote:
>>
>>> I found the answer by myself while I had finished writing the e-mail. The
>>> correct call url is loopback/ (in this case the command
>>> is originate loopback/fakecall 1000 ).  I'm sending the e-mail anyway for
>>> future reference (can't find any example of that anywhere). Is the project
>>> wiki accesible for anyone to contribute or do I have to ask for an
>>> authorization?
>>>
>>
>> All you need to do is sign up for a free account on the wiki and you can
>> start editing. It's a community resource and all FS users are invited to add
>> their respective knowledge.
>>
>> As for not finding what you were looking for, does this page not have it?
>> http://wiki.freeswitch.org/wiki/Loopback
>>
>> If not then please feel free to add to this page whatever your specific
>> scenario entails and give some examples.
>>
>> -MC
>>
>>
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Re: [Freeswitch-users] How can I join two freeswitch on two servers?

[Dan Le]

If you want FS server A to be able to call FS server B, you can set up a
user account in server B's FS directory configs, and then just treat server
B as a normal gateway by adding a gateway definition in server A. That will
allow you to route calls to server B from A; to do the reverse, just mirror
the configs the other direction.

On Mon, Jun 15, 2009 at 9:38 PM, Edmar Cruz  wrote:

>
> I like to connect two freeswitch, call each other, communicate and vice
> versa.
> Can you give me an example for that?
>
> Thanks
> --
> View this message in context:
> http://www.nabble.com/How-can-I-join-two-freeswitch-on-two-servers--tp24045824p24045824.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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Re: [Freeswitch-users] How to check Sofia status

[Dan Le]

I think he's asking if there's a way to check the status of a registered
user; if the user is idle versus in a call.
On Wed, Apr 22, 2009 at 12:38 PM, Michael Collins wrote:

> Could you add some more details to this question? I'm not sure I understand
> what you mean...
>
> Thanks,
> MC
>
> On Wed, Apr 22, 2009 at 12:58 AM, Baskar  wrote:
>
>> Hi,
>>
>> There is any api command to check the status of the extension whether the
>> agent is in ideal or in calling .
>>
>> Can any one asset me to solve the problem. Thanks in advance
>>
>>
>>
>>
>> --
>> Warm Regards,
>> N.Baskar
>>
>>
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Re: [Freeswitch-users] How to use variables in console/event socket

[Dan Le]

Or if you specifically want to avoid inputting the domain in an originate to
an internal profile, you can bind a sip profile to a domain
e.g.,  (in
sip_profiles/internal.xml)

then you can do: originate sofia/internal/1001 

Dan

On Tue, Apr 21, 2009 at 10:59 AM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:

> if you update to r13101 or better you can use the new expand function to do
> it
>
> expand originate sofia/internal/1001%${domain} 
>
>
> On Tue, Apr 21, 2009 at 8:27 AM, Brian West  wrote:
>
>> Variables aren't expanded on the command line.  You'll have to put exactly
>> what you want in.
>> /b
>>
>> On Apr 21, 2009, at 3:48 AM, Szymon Olko wrote:
>>
>> Substitution of $ sign is not understood for me. Is it possible to use
>> variables in that channels?
>> Variable is defined:
>>
>> freeswi...@vertux> global_getvar domain
>> API CALL [global_getvar(domain)] output:
>> 192.168.77.248
>>
>>
>>   Brian West
>> br...@freeswitch.org
>>
>> -- Meet us at ClueCon!  http://www.cluecon.com
>>
>>
>>
>>
>>
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>
>
> --
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>
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> ClueCon http://www.cluecon.com/
>
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>
> FreeSWITCH Developer Conference
> sip:8...@conference.freeswitch.org 
> iax:gu...@conference.freeswitch.org/888
> googletalk:conf+...@conference.freeswitch.org
> pstn:213-799-1400
>
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Re: [Freeswitch-users] Originate and Conference

[Dan Le]

I get similar behavior as Peter when trying to enter a locked conference.
If I am just dialing from a phone to a conference (on a dialplan), it will
properly lock me out. But if I do an originate command (originate
sofia/internal/1001 &conference(3000)), it will drop me into the conference,
even though it is suppose to be locked.

I am using the released 1.0.3 tag.


On Fri, Mar 27, 2009 at 4:27 AM, Peter P GMX  wrote:

> It's defined via XML-Curl, and manual dialling and transfering do
> trigger the same xml-curl request. This means that this conference
> number is not defined in the any xml conf file.
> If I transfer a call (without PIN) and then manually dial with another
> phone into this conf with PIN, both calls are in the same conference.
>
> I have SVN rev 12796.
>
>
> Best regards
> Peter
>
>
> Michael Collins schrieb:
> > 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
> >
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Re: [Freeswitch-users] SIP registration/retry/authorization problem

[Dan Le]

Thanks, wiki'd as:
http://wiki.freeswitch.org/wiki/Sofia#Forcing_SIP_profile_to_use_a_static_IP_address
Dan

On Wed, Mar 4, 2009 at 7:00 PM, Mathieu Rene  wrote:

> it auto restarts the profile when a network address change is detected.-
> Show quoted text -
>
> On 4-Mar-09, at 6:58 PM, Dan Le wrote:
>
> What does  do exactly? When do
> you need it?
> I'll wikify the response.
>
> Thanks,
> Dan
>
> On Tue, Feb 10, 2009 at 8:48 PM, Brian West  wrote:
>
>> I highly recommend you wipe the box/install and install from Scratch
>> using SVN trunk
>>
>>
>> /b
>> - Show quoted text -
>>
>> On Feb 10, 2009, at 7:43 PM, Jesse Peterson wrote:
>>
>> > I'm not able to find any documentation on this setting. I think it may
>> > be newer than my version of FreeSwitch (1.0). What does it do?
>> >
>> > Thanks,
>> > - Jesse
>>
>>
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Re: [Freeswitch-users] SIP registration/retry/authorization problem

[Dan Le]

What does  do exactly? When do you
need it?
I'll wikify the response.

Thanks,
Dan

On Tue, Feb 10, 2009 at 8:48 PM, Brian West  wrote:

> I highly recommend you wipe the box/install and install from Scratch
> using SVN trunk
>
>
> /b
> - Show quoted text -
>
> On Feb 10, 2009, at 7:43 PM, Jesse Peterson wrote:
>
> > I'm not able to find any documentation on this setting. I think it may
> > be newer than my version of FreeSwitch (1.0). What does it do?
> >
> > Thanks,
> > - Jesse
>
>
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Re: [Freeswitch-users] key tone trigger event during call

[Dan Le]

>From my understanding, I didn't think tone_detect detects DTMF since it's
dual frequencies, rather tone_detect detects single frequencies like fax
tones.

I would just run an IVR with a session.read or session.getDigits to collect
DTMF.

Dan

On Sat, Dec 6, 2008 at 1:19 PM, Frank @ Impact <[EMAIL PROTECTED]> wrote:

> Same thing with version 10640 build.
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Brian West
> Sent: Saturday, December 06, 2008 1:01 AM
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] key tone trigger event during call
>
> make current or install current svn on a different box.
>
> /b
>
> On Dec 5, 2008, at 7:09 PM, Frank @ Impact wrote:
>
> >
> > Ideas?  Am I doing something stupid or is tone_detect not just right
> > here?
>
>
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Re: [Freeswitch-users] Conference

[Dan Le]

Does the second call to 9840544078 have to come from 1002? You can always
just call 9840544078 and drop it into the conference directly:

api originate sofia/default/[EMAIL PROTECTED] &conference(3001)

or set the channel variable to simulate it coming from 1002

api originate {origination_caller_id_number=1002}sofia/default/
[EMAIL PROTECTED] &conference(3001)

Granted, not sure I completely understand your problem.

---

danule


On Wed, Nov 12, 2008 at 2:30 AM, Baskar <[EMAIL PROTECTED]> wrote:

> Hi,
>
> *Just i want to call the 1002==>9841799874 and i want to call another no
> 9840544078 and add in to the conference room  *
>
> *First method i tried :*
>
> api originate sofia/internal/1002%172.20.191.227 &bridge(sofia/default/
> [EMAIL PROTECTED]) >1002 call 9841799874
>
> api uuid_transfer 04120c08-19e7-4881-8873-0834a11524ff -both 3001
> >Both caller and extension will be transfer into conference room
>
> Then i call the another caller
>
> api originate sofia/internal/1002%172.20.191.227 &bridge(sofia/default/
> [EMAIL PROTECTED])> 1002 call another caller 9840544078
>
> api uuid_transfer c564f668-6778-4e78-85f6-a78c9c67cfa4 -bleg 3001
> >Transfer only the caller 9840544078 to conference room 3001
>
> while calling the second no 9840544078 the first line get hold
>
> after the transfer of bleg they all come in to the conference room but the
> first line caller is in hold so we cant able to unhold the call through api
> we have to manually go the soft phone and unhold the call .
>
> but i need the unhold should be done in api  what is the solution .
>
> *i have tried in another method also *
>
> api originate sofia/internal/1002%172.20.191.227 &bridge(sofia/default/
> [EMAIL PROTECTED])>1002 call 9841799874
>
> api uuid_transfer 04120c08-19e7-4881-8873-0834a11524ff -both 3001
> >Transfer only the caller 9841799874 to conference room 3001
>
> api uuid_kill 04120c08-19e7-4881-8873-0834a11524ff
>
> before i hangup  i have transfer both the caller in to conference room so
> when i hangup the extension why my extension got hangup
>
> Can any one suggest me how can i try the conference are else correct where
> i am wrong.
>
> --
> Warm Regards,
> N.Baskar
>
>
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