[Freeswitch-dev] [Freeswitch-users] C SAY API
Corelynx Admin
admin at corelynx.com
Sun Dec 17 23:29:16 EST 2006
Hi Kannaiyan,
As it sounds you have good exposure on Freeswitch, wondering if it can be used for heavy SIP to H323 translation on signal basis without the need of doing any media proxy. We are looking for a good alternative to Cisco 3875 if it be derived from a server based solution. Someone did tell me that Intel with its HMP card is able to deliver that at low cost but wanted to get more inputs which would be a better route and what will ensure quick expansion as required without having to invest in the hardware that much.
Rgds,
Corelynx Inc
----- Original Message -----
From: Kannaiyan Natesan
To: freeswitch-dev at lists.freeswitch.org
Sent: Sunday, December 17, 2006 4:57 AM
Subject: [Norton AntiSpam] [Freeswitch-dev] [Freeswitch-users] C SAY API
vset = LoadVoiceSet("antony","en");
do the operationg based upon this voiceset
FreeVoiceSet(vset);
This will be handy to handle mutliple language, multiple user customised voice prompts.
I'm sure this is lacking in the existing pbx structures.
The sound directory looks like,
sound
|
|---> sequence
| |
| |----> language
| |
| |-----> en
| | |----> voicemail.txt
| |-----> tamil
| |----> voicemail.txt
|
|->default
| |
| |-->language
| | |---> en
| | | |----> digits
| | | | |-------> 0 ... 9 .(format)
| | | |
| | | |----> Words
| | | |-------> you.format (you.wav, you.mp3)
| | | |-------> have.format
| | | ......... all possibles
| | |
| | |---> tamil
| (same structure as above)
|
|->antony
(same as above)
Sequence Directory:
It explains you what is the framing sequence of the file you want for the application. Sequence of the voice file is not same for all languages that will vary, so there is a definite need of this sequence(name it with a right word) so that it will be generic to all applications.
eg.,
[voicemail.txt] (forvoicemail app)
welcome=welcome to freeswitch voicemail system
in case if I use tamil language
eg.,
[voicemail.txt]
welcome=freeswitch kuralanjal amaippirkku varaverkirathu
I'm talking about this on each call basis, I mean it can take english call, tamil call, french call .... simultaneously.
Just my 2 cents.
Thanks.
Regards,
Kannaiyan
On 12/17/06, Nikolay Kolev < me at nikolay.com> wrote:
>
>
>
>
> The idea is great!
>
>
>
> OpenPBX seems to have one something similar, but it's unfortunately GPL.
>
>
>
> BTW, SSM_ITTERATED is misspelled. What would be the difference between
> SSM_ITERATED and SSM_PRONOUNCED?
>
>
>
> Nikolay Kolev
>
> ________________________________
>
>
> From: Anthony Minessale [mailto:anthmct at yahoo.com]
> Sent: Friday, December 15, 2006 3:09 PM
> To: freeswitch-dev at lists.freeswitch.org;
> freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-dev] C SAY API
>
>
>
>
>
> Hi,
>
>
>
>
>
> I'm working on a muiti-lingual say api for ivr programmers to be able
>
>
> to generate audio for saying numbers, counting items or expressing amounts
> of things such as currency. I'd like some input from anyone who has some
> ideas
>
>
> on how to make sure we cover all bases.
>
>
>
>
>
> The basic idea is that we will have an api where each new
> language/implementation would provide it's own version of this function.
>
>
>
>
>
> switch_status_t say(char *tosay,
> switch_say_gender_t gender,
> switch_say_type_t type,
> switch_say_method_t method,
> switch_input_callback_function_t
> dtmf_callback,
> void *buf,
> uint32_t buflen);
>
>
> The enums are described in
> http://www.freeswitch.org/eg/say_api.c
>
>
>
>
>
> the idea would be something like
>
>
>
>
>
> switch_say_handle_t *en;
>
>
>
>
>
> en = switch_say_load("en");
>
>
>
>
>
> switch_ivr_play_file(session, NULL, "/sounds/youhave.wav", NULL, NULL, NULL,
> 0);
>
>
> en->say("1000", SSG_ANY, SSM_PRONOUNCED, SST_MESSAGES, NULL, NULL, 0);
>
>
> switch_ivr_play_file(session, NULL, "/sounds/msgs.wav", NULL, NULL, NULL,
> 0);
>
>
>
>
>
>
> Thoughts?
>
>
>
>
>
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> JABBER: anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
>
>
> FreeSWITCH Developer Conference
> sip:888 at 66.250.68.194
> iax:guest at 66.250.68.194/888
> googletalk:freeswitch at gmail.com
> pstn:213-799-1400
>
>
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