Re: [Freeswitch-users] skill-based ACD

[Remko Kloosterman]

Hello Michael,

 

Do you still want to follow up on this? I'm having difficulty gathering
the old stuff in an understandable form. Also, it looks like the open
source ACD Spice Telephony by Andrew Thompson can do just what you might
need. 

 

Remko

 

Van: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] Namens Michael
Collins
Verzonden: donderdag 17 september 2009 22:58
Aan: freeswitch-users@lists.freeswitch.org
Onderwerp: Re: [Freeswitch-users] skill-based ACD

 

 

On Thu, Sep 17, 2009 at 12:13 PM, Remko Kloosterman
 wrote:

I have been working on several voice projects in the past with ACD
features, mostly based on TDM technology. It's all commercial stuff, but
I have the experience and I am willing to share that. If anyone wishes
to start such a development I'm sure I can dig up a functional model and
help with the design. 

 

I would like to see the functional model. That sounds interesting. We
could take it from there. Perhaps the FS community will have a few
members willing to help out.
-MC

 

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Re: [Freeswitch-users] skill-based ACD

[Remko Kloosterman]

This actually sounds very good Andrew. You even have an agent interface.

Do you have plans for a outbound campaign dialer? I know of a commercial
dialer that is good in it's predictive algotithm, but very bad when it
comes to campaign management.

-Oorspronkelijk bericht-
Van: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] Namens Andrew
Thompson
Verzonden: vrijdag 18 september 2009 20:56
Aan: freeswitch-users@lists.freeswitch.org
Onderwerp: Re: [Freeswitch-users] skill-based ACD

On Thu, Sep 17, 2009 at 11:20:22AM -0700, Michael Collins wrote:
> I was curious about this myself. Even if someone has built a non-free
> skills-based ACD using FS I'd like to know about it.
> -MC

I guess nobody paid any attention to my Cluecon presentation... :(

http://wiki.opencsm.org/wiki/index.php/Spice_Telephony is a skill-based
ACD that uses FS for its voice components. I havent pimped it here in
quite a while but here's some of its major features

* Skill based routing
* Priority Queues (instead of just FIFO)
* Multiple call types (voice, voicemail and email are currently
  supported, instant message support (via libpurple) is prototyped)
* Outbound call support (no autodialer though)
* Distributed system so you can aggregate multiple FS
  instances/locations into one big 'virtual' callcenter
* Web-based agent and administrative interface

There's quite a bit more, but that's the overview. The project is
finally approaching a 1.0 after over a year of development - I hope to
deploy it in production sometime around the end of this year or the
beginning of 2010 (replacing my previous custom asterisk solution).

You can grab the code at
http://git.opencsm.org/index.cgi/spice-telephony/ (you can browse or
git clone that URL). All you should need to run it is a modern erlang
release (R12B5 or newer) and ruby/rake to run the build.

Andrew

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Re: [Freeswitch-users] skill-based ACD

[Remko Kloosterman]

I have been working on several voice projects in the past with ACD
features, mostly based on TDM technology. It's all commercial stuff, but
I have the experience and I am willing to share that. If anyone wishes
to start such a development I'm sure I can dig up a functional model and
help with the design. 
 
Regards,
Remko



Van: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] Namens Christian
Jensen
Verzonden: donderdag 17 september 2009 20:57
Aan: freeswitch-users@lists.freeswitch.org
Onderwerp: Re: [Freeswitch-users] skill-based ACD



This would be a fantastic addition - my company is currently looking to
Asterisk as a potential candidate for this if FS can't do it.

 

I want FS to win of course :-)

 

Christian Jensen
Software Development Manager

Back Office



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: Thursday, September 17, 2009 11:20 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] skill-based ACD

 

 

On Sun, Sep 13, 2009 at 8:01 AM, mark morreny 
wrote:

Hello

 

Has any tried setting up an ACD based on skillset?  The current
out-of-box version of fifo does not seem to support acd based on agent
skillset.  Does anyone have any experience in doing it with some
external scripting using lua or javascript?  

 

I am interested in hearing how others may have done it as I am trying to
implement one myself.

 

thx,

 

mark

 


I was curious about this myself. Even if someone has built a non-free
skills-based ACD using FS I'd like to know about it.
-MC

 

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Re: [Freeswitch-users] ESL: DTMF event is not coming

[Remko Kloosterman]

I'm currently playing around with ESL myself. By default events are not
sent to the application. You should enable this by with for example the
"myevents" command. For a better understanding, try catching the IO with
"nc -l -p 8084" for outbound socket applications. See
http://wiki.freeswitch.org/wiki/Mod_event_socket for more info.
 
Remko



Van: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] Namens Michael
Gende
Verzonden: dinsdag 8 september 2009 5:28
Aan: freeswitch-users@lists.freeswitch.org
Onderwerp: Re: [Freeswitch-users] ESL: DTMF event is not coming


I'm pretty new at this, but please let me ask you a question: Is your
FreeSwitch running on a dual-homed host? 


On Mon, Sep 7, 2009 at 2:47 AM, Nagalenoj  wrote:



Dear friends,
  I am using freeswitch-1.0.4. When I execute the sample
script(/libs/esl/perl/server2.pl), it is not receiving the DTMF
events. When
I execute the same program in freeswitch-1.0.3, it's receiving
the event. Do
I miss something to configure/upgrade.?
  



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Re: [Freeswitch-users] ESL C questions

[Remko Kloosterman]

Thanks. 

Indeed it's a bit 'spartan' but it does describe the esl functions and 
structures.

 

Remko

 

Van: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] Namens João Mesquita
Verzonden: maandag 7 september 2009 19:21
Aan: freeswitch-users@lists.freeswitch.org
Onderwerp: Re: [Freeswitch-users] ESL C questions

 

Remko,

I wrote the documentation that is on docs.freeswitch.org

Take a look there, it is far from being complete but it might help.

jmesquita

On Mon, Sep 7, 2009 at 10:26 AM, Remko Kloosterman  
wrote:

Hi there,

 

I wonder, is ESL documentation available for C or does someone have something 
in draft? I'm trying to write an outbound socket application for some generic 
IVR features. I didn't find exactly that on the wiki except TODO J. The 
perl/ruby/javascript pages help a bit and the libs/esl source code provides 
examples that seem useful for trial and error, but I'd rather understand a bit 
more first.

 

Right now I have a socket server that forks a process, answers a call, 
generates beep and plays voice. How can I retrieve digits? Place an outbound 
call and bridge both legs or retrieve a cause if the call failed? Send/receive 
SIP INFO? Disconnect the call with some cause code? And all that (and some 
more) in C. Any help or pointers is appreciated.

 

Thanks,

Remko

 

 


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[Freeswitch-users] ESL C questions

[Remko Kloosterman]

Hi there,

 

I wonder, is ESL documentation available for C or does someone have
something in draft? I'm trying to write an outbound socket application
for some generic IVR features. I didn't find exactly that on the wiki
except TODO J. The perl/ruby/javascript pages help a bit and the
libs/esl source code provides examples that seem useful for trial and
error, but I'd rather understand a bit more first.

 

Right now I have a socket server that forks a process, answers a call,
generates beep and plays voice. How can I retrieve digits? Place an
outbound call and bridge both legs or retrieve a cause if the call
failed? Send/receive SIP INFO? Disconnect the call with some cause code?
And all that (and some more) in C. Any help or pointers is appreciated.

 

Thanks,

Remko

 

 

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Re: [Freeswitch-users] Which GSM gateway to buy?

[Remko Kloosterman]

I have used board-based voice solutions for ISDN/SS7 over the last 10 years and 
just moved to gateways and VoIP 2 years ago. I must say I've never been more 
happy. It scales excellent and is very stable, never had any problem and it's 
handling about 30.000 calls per day. In the past I've had a fair share of board 
issues, some crashes, scaling headache. Not totaly bad, but the gateway 
approach is much nicer. Check out audiocodes, I believe their mediant series 
supports GSM as well.
 
Remko



Van: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] Namens João Mesquita
Verzonden: woensdag 17 juni 2009 22:27
Aan: freeswitch-users@lists.freeswitch.org
Onderwerp: Re: [Freeswitch-users] Which GSM gateway to buy?


Guys, I was looking at the advantages and disadvantages of having a GSM gateway 
vs. a GSM board.

The conclusions I get are:

Board pros

1. Boards are able to get/send SMS without SIP "tricks"
2. You don't have to make a SIP call to check if channel is available and don't 
rely o SIP messages to get channel status
3. FS will be able to check for signal level on the board and fire events on 
pre-defined thresholds.
 
Gateway pros

1. I think of is the a GW can be used by more then one server, therefore, can 
have failover.
2. A GW is more scalable

It would be nice if you, that have already used GSM GWs in production, could 
comment on this.

Thanks,

jmesquita


On Wed, Jun 17, 2009 at 3:18 AM, NOx-WHV  wrote:



Hi,

look at www.kuhnt.com. It´s a german page. There you can find "Kontakt"
where you can ask for special requirements.

NOx




Diego Viola wrote:
>
> Hi everyone,
>
> Can you please recommend me some GSM gateway? I'm currently looking
> for a good one to buy... anyone have experience PORTech GSM gateways?
> Are they good?
>
> I also need it to work with FS, I'm also kinda new with VoIP hardware.
>
> Thanks,
>
> Diego
>
> ___
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>


--
View this message in context: 
http://www.nabble.com/Which-GSM-gateway-to-buy--tp24063401p24067617.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



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Re: [Freeswitch-users] DTMF recognition flaky

[Remko Kloosterman]

Hi Jay,

 

Did you make a wireshark trace yet? You should be able to find out
exactly what's going on there, which protocol is used, etc. We've had
our share of problems with DTMF over SIP trunks as well. Your problems
could also be related to timing issues introduced by multiple gateways.
Do you know some details on voicepulse's network? There's lots of
variations in implementation out there, unfortunately not always fully
compatible. 

 

Good luck,

Remko

 

 

Van: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] Namens Jay Austad
Verzonden: woensdag 6 mei 2009 20:57
Aan: freeswitch-users@lists.freeswitch.org
Onderwerp: Re: [Freeswitch-users] DTMF recognition flaky

 

I'm running 1.0.4pre3.  Haven't gotten a chance to upgrade to pre7 yet.


 

2833 is the default right?  I haven't changed anything.  I'm using
voicepulse for my SIP trunks.  Is there an option I can add to that
definition to force RFC2833?

 

--

jay austad  |  612.423.1433  |  aus...@signal15.com

 

 

 

 

On May 6, 2009, at 1:46 PM, Brian West wrote:





Well it depends.. first off are you doing inband dtmf or RFC2833?
Secondly what SVN rev are you running?

 

/b

 

On May 6, 2009, at 1:44 PM, Jay Austad wrote:





Using the default installation, I've noticed that when I (or someone  
else) calls in on my SIP trunk and keys in an extension, not all of  
the numbers are recognized unless they hold the key down for at least  
1/2 second to a second.

Is there a way to improve DTMF recognition so people can just type in  
stuff without having to hold the keys down?



 

Brian West

br...@freeswitch.org

 

-- Meet us at ClueCon!  http://www.cluecon.com 


 

 

 

 

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Re: [Freeswitch-users] VoiceXML

[Remko Kloosterman]

Excellent! I wasn't aware of their development progress. Let's check it
out.



Van: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] Namens EdPimentl
Verzonden: vrijdag 24 april 2009 15:10
Aan: freeswitch-users@lists.freeswitch.org
Onderwerp: Re: [Freeswitch-users] VoiceXML


Here is one editor to look into
http://www.eclipse.org/vtp/

-E
Gpro.ws

http://TwiTR.Me  (Cross Social Network Messaging Bus)
http://WatchNtweet.Me (Watch and Chat/Tweet) SocialTV
http://TwebEX.com (Twitter Based Online Web Conference Platform)
http://TweetUp.ws  (Twitter based  MeetUp service)


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Re: [Freeswitch-users] VoiceXML

[Remko Kloosterman]

For compliance it's needed to support it, but it's not actually needed
to make a working application. You can just use wave files like this:

  

  

  

Voxeo actually has a nice set of tutorials and reference material on it:
http://www.vxml.org/
  

-Oorspronkelijk bericht-
Van: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] Namens Michael
Jerris
Verzonden: woensdag 22 april 2009 7:12
Aan: freeswitch-users@lists.freeswitch.org
Onderwerp: Re: [Freeswitch-users] VoiceXML

You can use VXML without ASR, but ASR and TTS are both required parts of
the specs.

Mike

On Apr 21, 2009, at 10:41 PM, Brian West wrote:

> Isn't ASR optional in VXML?
>
> /b
>
> On Apr 21, 2009, at 6:15 PM, David Knell wrote:
>
>> Out of interest, is that using some RAD tool or coding directly in 
>> VoiceXML?  I ask because VoiceXML strikes me as being a bastard 
>> abomination of the highest order, whose sole saving grace is that 
>> it's a standardised bastard abomination.
>>
>> Or is Pocketsphinx the problem?


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Re: [Freeswitch-users] VoiceXML

[Remko Kloosterman]

I agree that the original commercial model used for VXML gets in the way
of it's own success. It's also focussed way too much on ASR/TTS. I think
we can all agree this technology is still a future promise, even after
10+ years. But technically VXML is an interesting concept, especially
together with related standards like ccxml
(http://en.wikipedia.org/wiki/Call_Control_eXtensible_Markup_Language).
The latter supports call transfer and conferencing. Every VoIP and TDM
product I've worked with in the past has it's own application interface
or is bound to an interface that's tailored for a special use (like
pbx). I've also used voxeo's callxml, that's implemented with the
CosmoCom ACD product. It's implementation is crappy, but the concept is
nice. The idea of separating callcontrol+media (voice/swithing core) and
application+ivr logic (vxml webserver cgi scripts) sounds rather
appealing to me.
 
Do you known the name of the free VXML editor/client that's out there?
 



Van: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] Namens Anthony
Minessale
Verzonden: woensdag 22 april 2009 3:58
Aan: freeswitch-users@lists.freeswitch.org
Onderwerp: Re: [Freeswitch-users] VoiceXML


I was initially turned off by VXML when it came out because the first
way they tried to make money off it was to 
sell the clients that let you build the actual xml.  I was not really
motivated to pay money to be able to 
just generate xml just so i could code a free server for it so I lost
interest.

I did hear there is now finally a free one out there. so that may make
it a little more reasonable.

I've commented in the past that I'm totally open to supporting VXML but
we have never had the public interest, time or resources
to work on it thus far.




On Tue, Apr 21, 2009 at 6:15 PM, David Knell  wrote:


On Tue, 2009-04-21 at 14:35 -0400, mszla...@aol.com wrote:
> Great Idea.
> Try setting up the exact same dialogue with say Voxeo's
VoiceXML
> system and then with Javascript/Lua and pocketsphinx. It's an
order of
> magnitude faster with VoiceXML.


Out of interest, is that using some RAD tool or coding directly
in
VoiceXML?  I ask because VoiceXML strikes me as being a bastard
abomination of the highest order, whose sole saving grace is
that
it's a standardised bastard abomination.

Or is Pocketsphinx the problem?

Cheers --

Dave





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-- 
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ClueCon http://www.cluecon.com/

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[Freeswitch-users] VoiceXML

[Remko Kloosterman]

Hi everyone,
 
Did someone ever try and implement a VXML interface on freeswitch? Or do
you think it's a good idea? Or not? Since it is an actual standard, I
guess there might be a market for application service providers.
 
Remko
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Re: [Freeswitch-users] [Freeswitch-dev] VMWare voice quality

[Remko Kloosterman]

Lot's of experience and suggestions here. Thanks.

I believe it should be theoretically possible to have blip-free RTP
streaming through the appliance. Most windows ethernet drivers allow for
QoS packet scheduling. If the VMware network bridge driver honors this
and syncs the buffers at 20ms frames (or whatever frame size applies)
you should be able to schale up a bit and maintain low jitter. 

Anyone knows how the VMware network bridge exactly works?


-Oorspronkelijk bericht-
Van: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] Namens Gregory
Boehnlein
Verzonden: donderdag 15 januari 2009 21:37
Aan: freeswitch-users@lists.freeswitch.org
Onderwerp: Re: [Freeswitch-users] [Freeswitch-dev] VMWare voice quality

> To the contrary, we have had quite good results in virtualized 
> environments and you don't really need timing that is that accurate to

> make it work.

If you don't handle RTP, I'm sure it is amazing. However, if you have to
do voicemail, stream audio from the server or do any kind of actual
time/latency/jitter sensitive processing, I don't care how much you tune
your hypervisor, it's never going to scale.

> We work quite well on amazon EC2 for example.  There are 2 issues I 
> know about with vmware, 1 is you need to set a setting on the host to 
> extend somewhat sane clocks being available, the second is I have seen

> issues with the bridged network adapter actually doubling up all 
> packets causing very strange issues, I suggest not using bridged 
> networking if you experience this.

I've not seen this behavior on Vmware ESX 3.5u2. Maybe an issue on
Vmware Server or Workstation?


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[Freeswitch-users] Vmware voice quality

[Remko Kloosterman]

Hello Ken, hello all,

I just read about the FreeSWITCH VMware applicance. I'm curious about
your experiences with the audio quality on VMWare, so here's a new
thread. 

I've installed freeswitch on VMware Server for Windows. The IVR audio
always plays choppy, while the server itself has no performance issues.
The same poor voice quality also goes for Asterisk or Yate, even on a
very fast VMware ESX system.

Did you experience the same and/or do you have pointers on how to
troubleshoot and fix this?

Thanks,

Remko

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[Freeswitch-users] VMWare voice quality

[Remko Kloosterman]

Hello Ken, hello all,

I just read about the FreeSWITCH VMware applicance. I'm curious about
your experiences with the audio quality on VMWare, so here's a new
thread. 

I've installed freeswitch on VMware Server for Windows. The IVR audio
always plays choppy, while the server itself has no performance issues.
The same poor voice quality also goes for Asterisk or Yate, even on a
very fast VMware ESX system.

Did you experience the same and/or do you have pointers on how to
troubleshoot and fix this?

Thanks,
Remko

-Oorspronkelijk bericht-
Van: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] Namens Ken Rice
Verzonden: donderdag 15 januari 2009 5:05
Aan: freeswitch-users@lists.freeswitch.org;
freeswitch-...@lists.freeswitch.org
Onderwerp: [Freeswitch-users] Announcing the FreeSWITCH Technology
PreviewVMWare Appliance.

Hey guys, 

I'm not trying to start 1 a day releases, Things just happened to fall
that way...

FreeSWITCH + Centos 5.2 rolled into a VMWare appliance that's compatible
with VMWare for Windows, VMWare Player, and VMWare Fusion on the Mac.

Couple of things you need to know. 1) This is a Centos 5.2 X86_64 that
was updated before we started testing it. 2) FreeSWITCH SVN Trunk is
installed and operational. See /etc/motd on the running image for all
the good information.

We'll be unvailing a wiki page for this shortly.

For now you can get the head start by downloading this at
http://files.freeswitch.org/FreeSWITCH-VM1.0.0.zip

Have fun guys!

Ken  



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