Re: [Freeswitch-users] Best way to determine if a bridge was successful in Lua

2009-05-22 Thread Matthew Fong
hrm...it's also seems to be that if my lua script looks like
session:execute(bridge, sofia/gateway/XXX/0X)
session:execute(bridge, sofia/gateway//XXX)

if the first bridge fails, the session is immediately hungup, even if
hangup_after_bridge is set to false...is this the intended behavior?

I'm not trying to setup failover--I know I can use | to setup a bridge
failover, but would like to retain use of the lua ivr script should a bridge
fail. If I want to redirect to a voicemail or recorded message, on bridge
fail, how can I do this? Thanks again.

--matt

On Thu, May 21, 2009 at 10:44 PM, Matthew Fong mattdf...@gmail.com wrote:

 I'm using a lua script to control an IVR, and would like to know how I can
 tell if a
 session:execute(bridge,sofia/gateway/blahblah);

 was successful or not

 it seems the response from session:execute is nil regardless if the bridge
 was successful or not

 whats the best way? Thanks

 --matt

___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


Re: [Freeswitch-users] Best way to determine if a bridge was successful in Lua

2009-05-22 Thread Matthew Fong
grr...continue_on_fail...ignore my ignorance ;)
but it would still be nice getting a response back from the session:execute
bridge

--matt

On Thu, May 21, 2009 at 11:09 PM, Matthew Fong mattdf...@gmail.com wrote:

 hrm...it's also seems to be that if my lua script looks like
 session:execute(bridge, sofia/gateway/XXX/0X)
 session:execute(bridge, sofia/gateway//XXX)

 if the first bridge fails, the session is immediately hungup, even if
 hangup_after_bridge is set to false...is this the intended behavior?

 I'm not trying to setup failover--I know I can use | to setup a bridge
 failover, but would like to retain use of the lua ivr script should a bridge
 fail. If I want to redirect to a voicemail or recorded message, on bridge
 fail, how can I do this? Thanks again.

 --matt

 On Thu, May 21, 2009 at 10:44 PM, Matthew Fong mattdf...@gmail.comwrote:

 I'm using a lua script to control an IVR, and would like to know how I can
 tell if a
 session:execute(bridge,sofia/gateway/blahblah);

 was successful or not

 it seems the response from session:execute is nil regardless if the bridge
 was successful or not

 whats the best way? Thanks

 --matt



___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


Re: [Freeswitch-users] Best way to determine if a bridge was successful in Lua

2009-05-22 Thread seven

would you like to try this?

bridge_hangup_cause = session:getVariable(bridge_hangup_cause) or  
session:getVariable(originate_disposition);
if  (bridge_hangup_cause == NORMAL_TEMPORARY_FAILURE or  
bridge_hangup_cause == NO_ROUTE_DESTINATION or bridge_hangup_cause  
== CALL_REJECTED)  then

-- blah...


On May 22, 2009, at 2:14 PM, Matthew Fong wrote:


grr...continue_on_fail...ignore my ignorance ;)

but it would still be nice getting a response back from the  
session:execute bridge


--matt

On Thu, May 21, 2009 at 11:09 PM, Matthew Fong mattdf...@gmail.com  
wrote:

hrm...it's also seems to be that if my lua script looks like

session:execute(bridge, sofia/gateway/XXX/0X)
session:execute(bridge, sofia/gateway//XXX)

if the first bridge fails, the session is immediately hungup, even  
if hangup_after_bridge is set to false...is this the intended  
behavior?


I'm not trying to setup failover--I know I can use | to setup a  
bridge failover, but would like to retain use of the lua ivr script  
should a bridge fail. If I want to redirect to a voicemail or  
recorded message, on bridge fail, how can I do this? Thanks again.


--matt

On Thu, May 21, 2009 at 10:44 PM, Matthew Fong mattdf...@gmail.com  
wrote:
I'm using a lua script to control an IVR, and would like to know how  
I can tell if a


session:execute(bridge,sofia/gateway/blahblah);

was successful or not

it seems the response from session:execute is nil regardless if the  
bridge was successful or not


whats the best way? Thanks

--matt


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


Re: [Freeswitch-users] Retrieving voicemail

2009-05-22 Thread Gavin Henry
2009/5/20 Michael Collins m...@freeswitch.org:
 Lars,

 Thanks for pointing this out. I will update the wiki. The new way to check
 voicemail is to dial 4000 and then enter your extension.

What version is that on?

I just followed
http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install and registered
with 1000/1234 and then dialed 1000 which put me straight into
voicemail. That link installs trunk, so should be the latest version
and default config?

Thanks.

___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


Re: [Freeswitch-users] text to speech IVRs and MOH

2009-05-22 Thread Saeed Ahmad
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
 Peter

 p...@privateconnect.com schrieb:
   I've spent the last 2-3 months on researching TTS and ASR for FS for a
  project.  Best TTS depends on what you consider important.  Also, how
  do you plan on using it.
 
  Here's some of the TTS engines I've run across with some pros/cons:
 
  Festivate Lite (flite)
  Pros:
  - Free (comes with FS)
  - simple to use
  - 16K voice sounds decent
  - Completely customizable
  Cons:
  - 8K voice sounds horrible over cell phone
 
  NeoSpeech (VoiceWare) (around $300/port for 1 voice + $75 each addl
 voice)
  Pros:
  - My selection for best soundig voices
  - Recently select by Stephen Hawkings for his voice (geek points!)
  - Lots of Languages supported
  - Free trial available
  Cons:
  - Custom C-Based API (FS interface coming soon)
  - Large file size (Engine + SDK + 1 Voice = 900MB)
  - Support is lacking (Company beed in Korean, time zone issues, etc)
 
  Nuance ($500/port for 1 voice)
  Pros:
  - Wide Variety of Products
  - Support MRCP
  - Supports ASR as well (add'l fees)
  - Excellent support
  - Free trial
  - Decent sounding voices at 8K and 16K
  - Wide range of tuning parameters
  Cons:
  - Pricey
  - Limited voice selection
  - Limited support for 64-bit linux
 
  ATT (NaturalVoice) (no pricing info available)
  Pros:
  - Big company (solid in marketplace)
  - Good suppport (user and developer)
  - ASP model means no software to maintain
  Cons:
  - ASP model incurs delay
  - Voices sound too digitized
  - Limited support for 64-bit linux
 
  Loquendo ($500/port for 1 voice + 15% addl voice)
  Pros:
  - Good sounding voices (almost as good as NeoSpeech)
  - Wide variety of languages
  - Excellent support
  - Has free 30 day trial
  - Supports MRCP
  - Support ASR and Voice Recognition as well. (add'l fees)
  - Small footprint ( 150MB)
  Cons:
  - Pricey
  - Complicated install process
  - Limited management/tuning capabilities
 
  In the end, it was down to NeoSpeech or Loquendo for our application.
   We are currently running tests with NeoSpeech and assuming all goes
  well, we will select them.  Though don't let that color your opinion
  too much after several focus groups we discovered the most important
  element in the equation is does your customer/boss like the sound of
  the voices, and that is a completely subjective decision.
 
 
  -pete
 
   Original Message 
  Subject: [Freeswitch-users] text to speech IVRs and MOH
  From: Saeed Ahmad saeedahmad1...@gmail.com
  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 speech IVR software
  which works OK with FS. i am using ATT site and for some IVRs i
  get sample rate errors. Also some resource to download more MOH
  wav files.
 
  Many thanks
 
 
  ___
  Freeswitch-users mailing list
  Freeswitch-users@lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:
 http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
 
  
  
  ___
  Freeswitch-users mailing list
  Freeswitch-users@lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
 

 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org

___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


Re: [Freeswitch-users] Help regarding configuration of FreeSWITCH to act as transparent proxy

2009-05-22 Thread Ognjen Seslija
Hello,

FS by design is B2BUA, and it cannot route INVITEs and other SIP methods. It
can however, bridge a-leg to b-leg with or w/o media and doing plenty other
cool stuff much better than commercial projects. I suggest joining us on irc
to detail your setup so we can help you.

Regards,
Ognjen (sekil on #freeswitch).

On Fri, May 22, 2009 at 7:24 AM, Rajagopal, Sridhar (Sridhar) 
sridh...@alcatel-lucent.com wrote:

  Hi all,

 I want to use FreeSWITCH as a SIP transparent proxy in session border
 controller application. Please let  me know the changes in  configuration
 files required to achieve this behaviour

 Thanks very much for the help.

 Regards,
 Sridhar


 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


Re: [Freeswitch-users] Help regarding configuration of FreeSWITCH to act as transparent proxy

2009-05-22 Thread Peter P GMX
This is also interesting for me, as I love freeswitch, and maintaining a
single platform is easier, than handling various different ones.
In the past years I did a couple of projects with OpenSER /openSIPS.
These projects comprised:

* registrar for the SIP user agents
* handle invite messages (+ ringing, bye, ok, etc also of course)
  between registered user agents and user agents at external domains
* rtp payload was a bit different from usual VoIP traffic (video
  parts, application sharing, file downloads etc.), but SDP was fine
  according to RFC, and OpenSER mediaproxy worked also
* handling of peer-to-peer presence (SUBSCRIBE, MEASSAGE, OPTIONS)
* The number of messages to handle was not that much (some thousand
  subs).

For my understanding this should also be possible with Freeswitch with
bypass_media. Right?

Best regards
Peter



Ognjen Seslija schrieb:
 Hello,

 FS by design is B2BUA, and it cannot route INVITEs and other SIP
 methods. It can however, bridge a-leg to b-leg with or w/o media and
 doing plenty other cool stuff much better than commercial projects. I
 suggest joining us on irc to detail your setup so we can help you.

 Regards,
 Ognjen (sekil on #freeswitch).

 On Fri, May 22, 2009 at 7:24 AM, Rajagopal, Sridhar (Sridhar)
 sridh...@alcatel-lucent.com mailto:sridh...@alcatel-lucent.com wrote:

 Hi all,
  
 I want to use FreeSWITCH as a SIP transparent proxy in session
 border controller application. Please let  me know the changes in 
 configuration files required to achieve this behaviour
  
 Thanks very much for the help.
  
 Regards,
 Sridhar
  

 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 mailto:Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org


 

 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org
   

___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


Re: [Freeswitch-users] calls appear to be dropping ... from landlines

2009-05-22 Thread Anthony Minessale
1) update to lastest trunk (you are at least 1000 revisions behind)
2) disable the presence debug in sofia.conf
3) enable sip trace instead sofia profile internal siptrace on
4) reproduce your problem.

Make sure you include more of the log from before the hangup happened.
The one you posted here is missing some of the info from the few seconds
prior but with the incomplete
info it looks like the other side sent a BYE ending the call.


On Thu, May 21, 2009 at 10:09 PM, Dale Trub dalet...@gmail.com wrote:

 Thanks Brian!  To answer your questions:
 Freeswitch svn revision: 12148
 Centos rev: 2.6.18-92.el5

 And apologies, actually I guess we're using g711 not 729.

 Jason:  I agree it would seem to be on the switch/telco side.  And, the
 telco says many other people are in the same set-up as us and don't have any
 issues, so they're insisting it's on our end.

 On Thu, May 21, 2009 at 7:28 PM, Brian West br...@freeswitch.org wrote:


 On May 21, 2009, at 9:15 PM, Dale Trub wrote:

 We're running FreeSwitch as part of a teleconferencing service, inside a
 telcom (so no
 internet latency/NAT issues) and using g.729


 So you're using g729 with conferences?

 We are receiving some complaints of dropped calls,
 including from landlines.   This means they join the conference, and x
 minutes in they simply drop.

 I know that cellphones tend to drop calls frequently, but landlines
 are pretty reliable, and we're hearing it a lot.  From the FreeSwitch side
 of things, it just
 looks like those callers hung up (but then dialed back in just a moment
 later).

 I'm attaching two different snippets of the FS log files where these
 issues are occurring.


 Next time please call them .txt because you cause extra work to have to
 open them otherwise.

 Does anyone have any recommendations about how to troubleshoot this?

 Any known issues/patches in FS that could be biting us?


 Depends you failed to include some very valid info such as what version or
 svn rev you're running and what linux distro.

 Is there some SIP logging we can do to debug?


 Yes covered on the wiki.
 http://wiki.freeswitch.org/wiki/Debugging_Freeswitch

 Are there any paid contractors avail who would have the expertise to look
 into this?


 email consult...@freeswitch.org

 Any help appreciated ... this is a major issue for us!

 Thanks much,

 -Dale


   Brian West
 br...@freeswitch.org

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





 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org



 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org




-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org
pstn:213-799-1400
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


Re: [Freeswitch-users] Secure RTP

2009-05-22 Thread Brian West


On May 22, 2009, at 12:47 AM, Jim Burke wrote:


Hey Brian,

Will have a look at ZRTP :)

Not sure I understand your comments regarding its all over once
receiving the 415 from the B party.  Is'nt that what parm
continue_on_fail does?  The fact that it sends the invite back out
sorta proves this.


The A-LEG has to hangup to re-enable SRTP it can't do it if it didn't  
invite with it in the first place.




The other point of interest here is that if you set action
application=export data=sip_secure_media=true/ before the first
bridge function it will include the security descriptions in the B leg
INVITE even when the A leg does not have them and the call will
succeed.  The B Eyebeam will show the locked padlock while A does not.


Make sure you do not answer the call before you do it.




From what I can see in code it is this guy that must stop it all from

happening.  TFLAG_SECURE  But I dont understand why :(


Again you have to invite to FS with crypto it can't magically cause  
crypto to work unless you initiate it with your first invite.




Regards,
Jim


Brian West
br...@freeswitch.org

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




___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


Re: [Freeswitch-users] Retrieving voicemail

2009-05-22 Thread Brian West
You have old configs with the old method in it if you had a  
previous install it won't overwrite the configs with new ones.


/b

On May 22, 2009, at 2:31 AM, Gavin Henry wrote:


What version is that on?

I just followed
http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install and registered
with 1000/1234 and then dialed 1000 which put me straight into
voicemail. That link installs trunk, so should be the latest version
and default config?

Thanks.


Brian West
br...@freeswitch.org

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




___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


Re: [Freeswitch-users] Retrieving voicemail

2009-05-22 Thread Gavin Henry
2009/5/22 Brian West br...@freeswitch.org:
 You have old configs with the old method in it if you had a previous
 install it won't overwrite the configs with new ones.


Ah, will re-install

___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


Re: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN Updates - Bootstrap Required

2009-05-22 Thread mszlazak

 What problems will a Windows user have when updating with Tortoise SVN?


 


 

-Original Message-
From: Michael Collins m...@freeswitch.org
To: freeswitch-users@lists.freeswitch.org 
freeswitch-users@lists.freeswitch.org; freeswitch-...@lists.freeswitch.org 
freeswitch-...@lists.freeswitch.org
Sent: Thu, 21 May 2009 8:54 pm
Subject: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN Updates - 
Bootstrap Required










FYI,

We just want to let everyone know that we have made a few updates that will 
require a rebootstrap. One of the key updates was a security fix for 
libsndfile. In this particular case it won't be possible simply to make 
current like you normally do. Here is a common set of commands for a typical 
Linux rebootstrap:


cd /usr/src/freeswitch.trunk
make clean
svn up
./bootstrap.sh
./configure
make install

NOTE: if you've got the libzrtp file and you've already run the buildzrtp.sh 
script then be sure to use ./configure --enable-zrtp in the above operation.


Thank you for your continued support of the FreeSWITCH project!

-Michael S Collins
http://www.cluecon.com





 





___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 

___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


Re: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN Updates - Bootstrap Required

2009-05-22 Thread Michael Collins
On Fri, May 22, 2009 at 7:58 AM, mszla...@aol.com wrote:

  What problems will a Windows user have when updating with Tortoise SVN?


I haven't had a chance to test it out but what I would do is update and then
rebuild solution and see how it goes. Let us know if you run into any
issues.

-MC
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


Re: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN Updates - Bootstrap Required

2009-05-22 Thread Brian West

Windows should have to clean the solution and rebuild

/b

On May 22, 2009, at 10:36 AM, Michael Collins wrote:




On Fri, May 22, 2009 at 7:58 AM, mszla...@aol.com wrote:
What problems will a Windows user have when updating with Tortoise  
SVN?


I haven't had a chance to test it out but what I would do is update  
and then rebuild solution and see how it goes. Let us know if you  
run into any issues.


-MC


Brian West
br...@freeswitch.org

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




___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


Re: [Freeswitch-users] Cool names for my VoIP company

2009-05-22 Thread SP
Dasbus

On Thu, May 21, 2009 at 23:26, Diego Viola diego.vi...@gmail.com wrote:
 Hey guys,

 I'm about to start my own ITSP with FreeSWITCH, and I'm looking some
 cool names for my VoIP company, if you know some please tell me :)

 Diego

 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org




-- 
Shannon

___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


Re: [Freeswitch-users] Cool names for my VoIP company

2009-05-22 Thread EdPimentl
VoiceCLOUD
CLOUDvoice
GlobalVoice
VoiceUP
voicEVERYthing
VoicEnterprise
S/IP (Services over IP)
GlobalSIP
VoiPLATFORM


Best regards,
-E
Gpro.ws
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


Re: [Freeswitch-users] Cool names for my VoIP company

2009-05-22 Thread Brian West

I say bkw_

On May 22, 2009, at 10:45 AM, SP wrote:


Dasbus


Brian West
br...@freeswitch.org

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




___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


Re: [Freeswitch-users] text to speech IVRs and MOH

2009-05-22 Thread mszlazak

 Neospeech does have the best voices. 


 I looked at Neospeech months ago and talked to a rep. 
Then he quoted me a price per port of over $1000.00. Looks like they really 
have done some big price adjustments.


 

-Original Message-
From: Saeed Ahmad saeedahmad1...@gmail.com
To: freeswitch-users@lists.freeswitch.org
Sent: Fri, 22 May 2009 2:01 am
Subject: Re: [Freeswitch-users] text to speech IVRs and MOH










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


Peter

p...@privateconnect.com schrieb:







 I've spent the last 2-3 months on researching TTS and ASR for FS for a
 project. ?Best TTS depends on what you consider important. ?Also, how
 do you plan on using it.

 Here's some of the TTS engines I've run across with some pros/cons:



 Festivate Lite (flite)
 Pros:
 - Free (comes with FS)
 - simple to use
 - 16K voice sounds decent
 - Completely customizable
 Cons:
 - 8K voice sounds horrible over cell phone



 NeoSpeech (VoiceWare) (around $300/port for 1 voice + $75 each addl voice)
 Pros:
 - My selection for best soundig voices
 - Recently select by Stephen Hawkings for his voice (geek points!)


 - Lots of Languages supported
 - Free trial available
 Cons:
 - Custom C-Based API (FS interface coming soon)
 - Large file size (Engine + SDK + 1 Voice = 900MB)
 - Support is lacking (Company beed in Korean, time zone issues, etc)



 Nuance ($500/port for 1 voice)
 Pros:
 - Wide Variety of Products
 - Support MRCP
 - Supports ASR as well (add'l fees)
 - Excellent support
 - Free trial
 - Decent sounding voices at 8K and 16K


 - Wide range of tuning parameters
 Cons:
 - Pricey
 - Limited voice selection
 - Limited support for 64-bit linux

 ATT (NaturalVoice) (no pricing info available)
 Pros:


 - Big company (solid in marketplace)
 - Good suppport (user and developer)
 - ASP model means no software to maintain
 Cons:
 - ASP model incurs delay
 - Voices sound too digitized


 - Limited support for 64-bit linux

 Loquendo ($500/port for 1 voice + 15% addl voice)
 Pros:
 - Good sounding voices (almost as good as NeoSpeech)
 - Wide variety of languages

 - Excellent support

 - Has free 30 day trial
 - Supports MRCP
 - Support ASR and Voice Recognition as well. (add'l fees)
 - Small footprint ( 150MB)
 Cons:
 - Pricey
 - Complicated install process


 - Limited management/tuning capabilities

 In the end, it was down to NeoSpeech or Loquendo for our application.
 ?We are currently running tests with NeoSpeech and assuming all goes
 well, we will select them. ?Though don't let that color your opinion


 too much after several focus groups we discovered the most important
 element in the equation is does your customer/boss like the sound of
 the voices, and that is a completely subjective decision.




 -pete

 ? ?  Original Message 
 ? ? Subject: [Freeswitch-users] text to speech IVRs and MOH
 ? ? From: Saeed Ahmad saeedahmad1...@gmail.com


 ? ? 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 speech IVR software


 ? ? which works OK with FS. i am using ATT site and for some IVRs i
 ? ? get sample rate errors. Also some resource to download more MOH
 ? ? wav files.

 ? ? Many thanks
 ? ? 


 ? ? ___
 ? ? Freeswitch-users mailing list
 ? ? Freeswitch-users@lists.freeswitch.org
 ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users


 ? ? UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 ? ? http://www.freeswitch.org





 








 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org


 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users


 http://www.freeswitch.org


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org


http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users


http://www.freeswitch.org








 





___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



 

___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

Re: [Freeswitch-users] text to speech IVRs and MOH

2009-05-22 Thread pete
Just to followup on Cepstral. I used Cepstral for an Asterisk project a while back. So my information may be dated. I had not considered the for this project based on quality of voices.Pros:- Cheaper than other solutions ($50/port, $30/voice)- Well documented- Good Support- Supports MRCP- Has ASP ModelCons:- Voices sounded a bit too digital
- Limited Languages available- Somewhat difficult install- Lacking a wide range of tuning options


 Original Message 
Subject: Re: [Freeswitch-users] text to speech IVRs and MOH
From: Saeed Ahmad saeedahmad1...@gmail.com
Date: Fri, May 22, 2009 2:01 am
To: 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:I've spent the last 2-3 months on researching TTS and ASR for FS for a project. Best TTS depends on what you consider important. Also, how do you plan on using it. Here's some of the TTS engines I've run across with some pros/cons:  Festivate Lite (flite) Pros: - Free (comes with FS) - simple to use - 16K voice sounds decent - Completely customizable Cons: - 8K voice sounds horrible over cell phone  NeoSpeech (VoiceWare) (around $300/port for 1 voice + $75 each addl voice) Pros: - My selection for best soundig voices - Recently select by Stephen Hawkings for his voice (geek points!)  - Lots of Languages supported - Free trial available Cons: - Custom C-Based API (FS interface coming soon) - Large file size (Engine + SDK + 1 Voice = 900MB) - Support is lacking (Company beed in Korean, time zone issues, etc)  Nuance ($500/port for 1 voice) Pros: - Wide Variety of Products - Support MRCP - Supports ASR as well (add'l fees) - Excellent support - Free trial - Decent sounding voices at 8K and 16K  - Wide range of tuning parameters Cons: - Pricey - Limited voice selection - Limited support for 64-bit linux ATT (NaturalVoice) (no pricing info available) Pros:  - Big company (solid in marketplace) - Good suppport (user and developer) - ASP model means no software to maintain Cons: - ASP model incurs delay - Voices sound too digitized  - Limited support for 64-bit linux Loquendo ($500/port for 1 voice + 15% addl voice) Pros: - Good sounding voices (almost as good as NeoSpeech) - Wide variety of languages  - Excellent support  - Has free 30 day trial - Supports MRCP - Support ASR and Voice Recognition as well. (add'l fees) - Small footprint ( 150MB) Cons: - Pricey - Complicated install process  - Limited management/tuning capabilities In the end, it was down to NeoSpeech or Loquendo for our application. We are currently running tests with NeoSpeech and assuming all goes well, we will select them. Though don't let that color your opinion  too much after several "focus groups" we discovered the most important element in the equation is does your customer/boss like the sound of the voices, and that is a completely subjective decision.  -pete    Original Message    Subject: [Freeswitch-users] text to speech IVRs and MOH   From: Saeed Ahmad saeedahmad1...@gmail.comDate: Tue, May 19, 2009 12:40 am   To: freeswitch-users@lists.freeswitch.org   Hi all,   Could you guys recommend me any online text to speech IVR softwarewhich works OK with FS. i am using ATT site and for some IVRs i   get sample rate errors. Also some resource to download more MOH   wav files.   Many thanks   ___   Freeswitch-users mailing list   Freeswitch-users@lists.freeswitch.org   http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users   http://www.freeswitch.org  ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users  http://www.freeswitch.org___Freeswitch-users mailing listFreeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

Re: [Freeswitch-users] Cool names for my VoIP company

2009-05-22 Thread Nandy Dagondon
how about InterTalk or InterMedia?

-nandy
===
LanVox Systems
Lapulapu City, Philippines 6015
Mobile: +63-920-6373450
Phone: +63-32-3401807
USA:   +1-360-8122281
http://sites.google.com/site/lanvoxphils



On Fri, May 22, 2009 at 11:52 PM, Brian West br...@freeswitch.org wrote:

 I say bkw_
 On May 22, 2009, at 10:45 AM, SP wrote:

 Dasbus


 Brian West
 br...@freeswitch.org

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





 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


Re: [Freeswitch-users] Cool names for my VoIP company

2009-05-22 Thread dujinfang
Voila itself is a good name. ;)

VoilaVoIP

On May 22, 2009, at 12:26 PM, Diego Viola wrote:

 Hey guys,

 I'm about to start my own ITSP with FreeSWITCH, and I'm looking some
 cool names for my VoIP company, if you know some please tell me :)

 Diego

 ___
 Freeswitch-users mailing list
 Freeswitch-users@lists.freeswitch.org
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org


___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


Re: [Freeswitch-users] Cool names for my VoIP company

2009-05-22 Thread Brian West

How about larynxvoip
/b

On May 22, 2009, at 7:10 PM, dujinfang wrote:


Voila itself is a good name. ;)

VoilaVoIP


Brian West
br...@freeswitch.org

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




___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org