[asterisk-users] Cepstral, Swift and Asterisk 13
Anyone here have a working app_swift with Asterisk 13? I purchased my licenses and followed their install procedure but I do not get any audio when I dial a test. Stranger still is that I can get audio on a softphone (Bria) but nowhere else. I have tried several desk phones and softphones and only Bria can get some distorded audio, all others are just silent. I get no error on the console so everything seems to be installed properly. Any pointers? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)8116-9161 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral voice quality not good
Actually it is bad only when received on cell phones. Today I listened to the same voices on a Cisco 7942 and they were great. I actually enjoyed listening to them. Not bad on X-Lite either. Previously I was mostly listening to them only through cell phones. So it means it is because of the transcodings at cell phone providers' ends. Bad though because many customers use cell phones exclusively. Maybe if I convert them to gsm format before playing, they'll play better, but will add delay and additional processing because they are converted and played in real time. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-23 6:24 PM, Zeeshan Zakaria zisha...@gmail.com wrote: I am using app_swift. As a side note, demo on their website also generates sounds which at places sounds like robotic. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-23 6:03 PM, Darren Sessions dmsessi...@gmail.com wrote: Are you using app_swift or wav files? On Oct 23, 2010, at 5:26 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Hello list, I hav... -- _ -- Bandwidth and ... -- _ -- Bandwidth and Colocation P... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral voice quality
Do you recommend using wav files instead? Will there be any downside of using wav? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -- _ -- Bandwidth and Colocatio... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral voice quality
Well, the downside to wav files is the disk i/o. Asterisk will and does translate the audio frames from ulaw to whatever other codec. Sent from my iPhone On Oct 24, 2010, at 9:42 AM, Zeeshan Zakaria zisha...@gmail.com wrote: Do you recommend using wav files instead? Will there be any downside of using wav? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -- _ -- Bandwidth and Colocatio... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral voice quality not good
I fiddled with the demo version of swift a year or so ago and I had better sound quality if I used the non-8khz versions and had app_swift or asterisk convert it for me (not sure, giving app_swift a regular version seemed to JustWork(tm) On Sat, Oct 23, 2010 at 3:24 PM, Zeeshan Zakaria zisha...@gmail.com wrote: I am using app_swift. As a side note, demo on their website also generates sounds which at places sounds like robotic. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-23 6:03 PM, Darren Sessions dmsessi...@gmail.com wrote: Are you using app_swift or wav files? On Oct 23, 2010, at 5:26 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Hello list, I hav... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cepstral voice quality not good
Hello list, I have been using Cepstral's 8KHz voices for my text-to-speech service for some time now, and have been noticing that the voice quality is really poor, doesn't matter what phrase I give it to convert. None of the other 8KHz voices I have ever used were this bad. It doesn't seem good enough system to be used in a commercial system. Is there any better quality text-to-voice engine? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral voice quality not good
Are you using app_swift or wav files? On Oct 23, 2010, at 5:26 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Hello list, I have been using Cepstral's 8KHz voices for my text-to-speech service for some time now, and have been noticing that the voice quality is really poor, doesn't matter what phrase I give it to convert. None of the other 8KHz voices I have ever used were this bad. It doesn't seem good enough system to be used in a commercial system. Is there any better quality text-to-voice engine? Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral voice quality not good
I am using app_swift. As a side note, demo on their website also generates sounds which at places sounds like robotic. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-10-23 6:03 PM, Darren Sessions dmsessi...@gmail.com wrote: Are you using app_swift or wav files? On Oct 23, 2010, at 5:26 PM, Zeeshan Zakaria zisha...@gmail.com wrote: Hello list, I hav... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
Somewhat off-topic, but I'll mention briefly that it's a multi-city service and you can get more info at http://www.trafficondemand.ca/ I believe that it's still considered beta for non-Toronto. You have Kitchener/Waterloo! Yay dials Oh. No traffic. Boo-urns. Hehe...working on it ;) I'd definitely like to know when you start populating the traffic part of K/W (and separate out london, it's a poor choice to group. Kitchener/Wwaterloo/Cambridge sure... but London? That's a common Torontonian thing to do. :-) Agreed. I advised the client against that, during design, but here we are. Hopefully he requests us to change this soon. On an unrelated note, I always find the Toronto is the centre of the universe attitude quite amusing. Some clients who call us for DSL qualifications, when asked Where are you located? respond with Bathurst Shephard. No sir, what city and province? -- Erik Caneris Tel: 647-723-6365 Fax: 647-723-5365 Toll-free: 1-866-827-0021 www.caneris.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
On December 2, 2008 07:55:00 pm Erik (Caneris) wrote: Nuance would say no :) I'd say maybe. Call up +14164854854, it's a recent project we did for a That's pretty cool! Is there any SIP or IAX access to this (aside from dialing a POTS number) ? -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival (MRCP)
John: However, that doesn't mean that it shouldn't be implemented. This is an area in which I think there is a disproportionate amount of non- discussion, since many people who would use or be interested in MRCP simply don't participate in the Asterisk project because it doesn't meet their needs out of the gate. Therefore, we see few people asking for it, in a self-fulfilling loop. Is MRCP something that is significantly lacking in Asterisk? Is it a difficult protocol to implement? Is there anyone here on -dev with the experience to do it? I don't know whether it's significantly lacking nor how difficult it is to implement, but it's certainly nice to have. It would increase the appeal of Asterisk to those used to working with MRCP-compatible resources in other platforms. That said, it can be argued that it's best to keep Asterisk simple and free of extra features. If its core purpose does not consist of interfacing with ASR and TTS engines, then some would argue that it's best to keep such features to a separate platform. Regards, -- Erik Caneris Tel: 647-723-6365 Fax: 647-723-5365 Toll-free: 1-866-827-0021 www.caneris.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
Thanks. Unfortunately no SIP/IAX access at this time, only by dialing one of the TNs. However, I'll bring it up with the client and see if they'd want us to configure that. Somewhat off-topic, but I'll mention briefly that it's a multi-city service and you can get more info at http://www.trafficondemand.ca/ I believe that it's still considered beta for non-Toronto. Regards, -- Erik Caneris Tel: 647-723-6365 Fax: 647-723-5365 Toll-free: 1-866-827-0021 www.caneris.com From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith (lists) [EMAIL PROTECTED] Sent: Thursday, December 04, 2008 10:43 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] cepstral vs festival On December 2, 2008 07:55:00 pm Erik (Caneris) wrote: Nuance would say no :) I'd say maybe. Call up +14164854854, it's a recent project we did for a That's pretty cool! Is there any SIP or IAX access to this (aside from dialing a POTS number) ? -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
On December 4, 2008 02:14:52 pm Erik (Caneris) wrote: Thanks. Unfortunately no SIP/IAX access at this time, only by dialing one of the TNs. However, I'll bring it up with the client and see if they'd want us to configure that. Definitely would be cool, you don't lose any ad revenue and I don't have to use up my minutes. Somewhat off-topic, but I'll mention briefly that it's a multi-city service and you can get more info at http://www.trafficondemand.ca/ I believe that it's still considered beta for non-Toronto. You have Kitchener/Waterloo! Yay dials Oh. No traffic. Boo-urns. I'd definitely like to know when you start populating the traffic part of K/W (and separate out london, it's a poor choice to group. Kitchener/Wwaterloo/Cambridge sure... but London? That's a common Torontonian thing to do. :-) -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival (MRCP)
On Dec 2, 2008, at 6:55 PM, Erik (Caneris) wrote: Erik - Have you found RealSpeak to be worth the cost? Actually my last note was probably a bit misleading because in the particular cases I mentioned RealSpeak, the platform wasn't Asterisk and Cepstral wasn't even on the radar. Can Cepstral, with the hourly $ spent on tuning, be made to be a reasonable substitute? Nuance would say no :) Of course, and perhaps they're right in some circumstances. But I don't think I'd be able to predict in what percentage of cases that's true. I'd say maybe. Call up +14164854854, it's a recent project we did for a client using Asterisk, Cepstral, and a lot of custom code. It's a free phone-in service that allows folks to get local traffic, weather, news, commuter transit, border crossing wait times, and more. There's obviously quite a bit of domain-specific, dynamic, constantly changing text, so this is certainly an example of pushing it to the max. Just think of all the street names it has the potential to mispronounce. It's a work in progress, but it's very promising. Definitely an example of a lot of hourly $ spent on tuning as you put it. Sounds decent. Some inter-word delays might be in order, but I'm sure that's how you're earning your keep. My results: The RealSpeak sample was more clear than the Cepstral. Depends on what you mean by more clear. As Brent Davidson mentions, make sure you're comparing 8khz to 8khz, or similar. If you mean it pronounces things better, then I agree. Of course, my test was hardly scientific. But I re-tested at 8khz for both voices, and both myself and someone else in the room (a non- expert) were not overwhelmed with the quality difference between the two voices. Totally subjective, but an apples-to-apples comparison. That being said, I'd really be interested in hearing if anyone has done a RealSpeak-to-Asterisk conduit. I wasn't able to quickly uncover how they interact with third-party systems - is it VoIP? A C library? Some sort of HTTP socket? The more methods we can get working with Asterisk, the better, because not every implementation of a voice system has the same requirements... MRCP is the standard for interfacing with ASR and TTS engines (including RealSpeak) in other platforms. Brief Googling reveals a previous flame war on asterisk-dev regarding MRCP. I have no idea if it's implemented in Asterisk now. No, it is not currently implemented. Note, though, that someone in another post mentioned that they had built an app_realspeak, and I'll try to follow up with that. However, that doesn't mean that it shouldn't be implemented. This is an area in which I think there is a disproportionate amount of non- discussion, since many people who would use or be interested in MRCP simply don't participate in the Asterisk project because it doesn't meet their needs out of the gate. Therefore, we see few people asking for it, in a self-fulfilling loop. Is MRCP something that is significantly lacking in Asterisk? Is it a difficult protocol to implement? Is there anyone here on -dev with the experience to do it? JT --- John Todd [EMAIL PROTECTED]+1-256-428-6083 Asterisk Open Source Community Director ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cepstral vs festival
I'm about to begin working on an ivr project to do database backed scheduling. I would like to use text to speech in some places. What are the differences in using festival vs. Cepstral? How are they similar, how are they different? Is one really better than the other? How and Why? Thanks, Eric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
Which non-english language do you have in mind ? Both should differ on this. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
In my experience cepstral has always had much nicer sounding voices, but I haven't tinkered too much with either. There is a reason one is pay and one free though J I believe cepstral is still offering demo's, I'd download each and see which one gives you the performance you're looking for. Thanks, Matt G : http://www.voipphreak.ca http://www.voipphreak.ca : http://www.ratemydialplan.com http://www.ratemydialplan.com : http://www.asterisk-jobs.com http://www.asterisk-jobs.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Fort Sent: Tuesday, December 02, 2008 3:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] cepstral vs festival I'm about to begin working on an ivr project to do database backed scheduling. I would like to use text to speech in some places. What are the differences in using festival vs. Cepstral? How are they similar, how are they different? Is one really better than the other? How and Why? Thanks, Eric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
Festival is a free voice that sounds like a machine. Cepstral is a fee based human voice ($30 USD per voice per CPU). They are similar in that they both produce mechanically timed output. IMO, you should use festival if this isn't a customer based interface. If it is a CBI, use cepstral and if you don't like it, recreate the wav files it plays (The English language is only based on about 1700 sounds). Cepstral is your choice if your IVR is going to be asterisk interlaced since all asterisk voices are Cepstral Allison out of the can. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Fort Sent: Tuesday, December 02, 2008 2:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] cepstral vs festival I'm about to begin working on an ivr project to do database backed scheduling. I would like to use text to speech in some places. What are the differences in using festival vs. Cepstral? How are they similar, how are they different? Is one really better than the other? How and Why? Thanks, Eric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
Festival sucks. Cepstral sucks less. The End. In my experience, it depends on the specific app, who's paying, and who's going to be the victim, err...user listening to it. This is the difference between domain/context specific phrases/words to pronounce vs. general stuff, a client on a tight budget or not, the users being internal vs. customers/public, and so on. Cepstral is a $30 TTS engine. It's not too bad, but you'll find mostly things like Realspeak deployed in large scale professional deployments, such as those used by the big boys, telcos/banks/airlines. We deployed Cepstral recently for a client, for a phone-in service used by the general public, and I can tell you that there was quite a bit of work in teaching it with SSML how to pronounce stuff. Again, it really depends on your specific situation. You should definitely try out those two at least and also ensure that the client/stakeholders are aware of limitations. There's a certain expectation of it will speak perfectly these days, followed by disappointment and blame when reality hits them. Regards, -- Erik Caneris Tel: 647-723-6365 Fax: 647-723-5365 Toll-free: 1-866-827-0021 www.caneris.com From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Eric Fort [EMAIL PROTECTED] Sent: Tuesday, December 02, 2008 3:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] cepstral vs festival I'm about to begin working on an ivr project to do database backed scheduling. I would like to use text to speech in some places. What are the differences in using festival vs. Cepstral? How are they similar, how are they different? Is one really better than the other? How and Why? Thanks, Eric ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Fort Sent: Tuesday, December 02, 2008 3:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] cepstral vs festival I'm about to begin working on an ivr project to do database backed scheduling. I would like to use text to speech in some places. What are the differences in using festival vs. Cepstral? How are they similar, how are they different? Is one really better than the other? How and Why? On Tue, 2 Dec 2008, Matt Gibson wrote: In my experience cepstral has always had much nicer sounding voices, but I haven't tinkered too much with either. There is a reason one is pay and one free though J I believe cepstral is still offering demo's, I'd download each and see which one gives you the performance you're looking for. Way back in the day, festival was awful and Cepstral as almost acceptable. Now, especially with their Allison font, Cepstral is good enough than you can't always tell the difference -- even without using their markup language. The fit with the live Allison's prompts included with Asterisk is great. It's fantastic for demos. You can refine the wording of your prompts before committing to live talent. You may decide that the tts prompts are good enough. I invoke swift (Cepstral's command line tts tool) to create my prompts from my makefile so it's easy to make changes and everything is documented. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
On Dec 2, 2008, at 9:41 AM, Erik (Caneris) wrote: Festival sucks. Cepstral sucks less. The End. In my experience, it depends on the specific app, who's paying, and who's going to be the victim, err...user listening to it. This is the difference between domain/context specific phrases/words to pronounce vs. general stuff, a client on a tight budget or not, the users being internal vs. customers/public, and so on. Cepstral is a $30 TTS engine. It's not too bad, but you'll find mostly things like Realspeak deployed in large scale professional deployments, such as those used by the big boys, telcos/banks/ airlines. We deployed Cepstral recently for a client, for a phone-in service used by the general public, and I can tell you that there was quite a bit of work in teaching it with SSML how to pronounce stuff. Again, it really depends on your specific situation. You should definitely try out those two at least and also ensure that the client/stakeholders are aware of limitations. There's a certain expectation of it will speak perfectly these days, followed by disappointment and blame when reality hits them. Regards, -- Erik Caneris Tel: 647-723-6365 Fax: 647-723-5365 Toll-free: 1-866-827-0021 www.caneris.com Erik - Have you found RealSpeak to be worth the cost? Can Cepstral, with the hourly $ spent on tuning, be made to be a reasonable substitute? It's been a while since I did a head-to-head comparison between Cepstral and (anything else) so I did a quick demo of the RealSpeak Host-based telecom app: http://www.nuance.com/realspeak/demo/ (contact data required) and the Cepstral demo: http://www.cepstral.com/demos/ I used the Jill (default - 8khz) for RealSpeak and Allison (default) for the tests, and played back the same phrase: Congratulations. You have successfully installed and executed the Asterisk open source PBX. My results: The RealSpeak sample was more clear than the Cepstral. But by how much? I should probably test with more than just that one phrase, but I can't say I'd prefer RealSpeak significantly over Cepstral in this extremely limited case. Does RealSpeak get better long-term test results and comprehension/retention? I know that Cepstral is $50/port - the RealSpeak pricing is un-findable, which tells me that it's significantly higher than Cepstral. (Personal peeve: at least put your list pricing on the website! grumble) That being said, I'd really be interested in hearing if anyone has done a RealSpeak-to-Asterisk conduit. I wasn't able to quickly uncover how they interact with third-party systems - is it VoIP? A C library? Some sort of HTTP socket? The more methods we can get working with Asterisk, the better, because not every implementation of a voice system has the same requirements... JT --- John Todd [EMAIL PROTECTED]+1-256-428-6083 Asterisk Open Source Community Director ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
John Todd wrote: Erik - Have you found RealSpeak to be worth the cost? Can Cepstral, with the hourly $ spent on tuning, be made to be a reasonable substitute? It's been a while since I did a head-to-head comparison between Cepstral and (anything else) so I did a quick demo of the RealSpeak Host-based telecom app: http://www.nuance.com/realspeak/demo/ (contact data required) and the Cepstral demo: http://www.cepstral.com/demos/ I used the Jill (default - 8khz) for RealSpeak and Allison (default) for the tests, and played back the same phrase: Congratulations. You have successfully installed and executed the Asterisk open source PBX. My results: The RealSpeak sample was more clear than the Cepstral. But by how much? I should probably test with more than just that one phrase, but I can't say I'd prefer RealSpeak significantly over Cepstral in this extremely limited case. Does RealSpeak get better long-term test results and comprehension/retention? I know that Cepstral is $50/port - the RealSpeak pricing is un-findable, which tells me that it's significantly higher than Cepstral. (Personal peeve: at least put your list pricing on the website! grumble) That being said, I'd really be interested in hearing if anyone has done a RealSpeak-to-Asterisk conduit. I wasn't able to quickly uncover how they interact with third-party systems - is it VoIP? A C library? Some sort of HTTP socket? The more methods we can get working with Asterisk, the better, because not every implementation of a voice system has the same requirements... JT --- John Todd [EMAIL PROTECTED]+1-256-428-6083 Asterisk Open Source Community Director This may not be a perfectly fair comparison. Looks like you're comparing the RealSpeak 8khz voice to the Cepstral default Allison which is NOT 8khz. If you look on the Cepstral site you'll see Desktop Voices and Telephony Voices. The Cepstral Telephony voices are 8khz, and I suspect their quality is on par with RealSpeak. I recently licensed the Allison-8Khz voice for some of the admin functions on my companies phone systems where I didn't want to record prompts and Flite was too robotic sounding. The Allison-8khz voice is virtually indistinguishable from the pre-recorded Allison prompts, except for maybe some minor differences in inflection. I was thoroughly impressed with the quality though. For the most part it sounds like you've hired Allison to record custom prompts. The Allison Desktop voice is OK, but sounds sort of like Allison is taking through a spinning fan blade. When you're doing TTS comparisons be sure you're comparing apples to apples and not peaches to apricots. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 John Todd a écrit : My results: The RealSpeak sample was more clear than the Cepstral. But by how much? I should probably test with more than just that one phrase, but I can't say I'd prefer RealSpeak significantly over Cepstral in this extremely limited case. Does RealSpeak get better long-term test results and comprehension/retention? I know that Cepstral is $50/port - the RealSpeak pricing is un-findable, which tells me that it's significantly higher than Cepstral. (Personal peeve: at least put your list pricing on the website! grumble) For French language, I find the quality of RealSpeak to be very good. Festival was unusable (for French); I tried Cepstral but was deceived. The price of RealSpeak is not far from an order of magnitude higher compared to Cepstral. That being said, I'd really be interested in hearing if anyone has done a RealSpeak-to-Asterisk conduit. I wasn't able to quickly uncover how they interact with third-party systems - is it VoIP? A C library? Some sort of HTTP socket? The more methods we can get working with Asterisk, the better, because not every implementation of a voice system has the same requirements... That's a C library. I bought RealSpeak SDK, and developed app_realspeak for Asterisk (1.2, then ported to 1.4). I've been using it since 2005 for my IVR projects, including telcos/banks/airlines :) Regards, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAkk10kAACgkQuu7Rv+oOo/gK2ACfXedtJ8k7cmVRpOqTU+rYpbVy PcIAnjbXbDPuicE29673TQY3CritOksQ =vvB7 -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 This has been an interesting discussion about cepstral. My question is why it doesn't appear to be available for 1.6 yet? This thread has piqued my interest in the product but a visit to Digium's website seems to point to it being a product for Asterisk 1.6. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJNdMwCFu3bIiwtTARAqqQAJ9mXLMyUCzI+UCiF3/1j4kuGE32ewCgpS2r 8IwCpap3Q1puuP4LZScVV00= =4Cdn -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
Jean-Denis Girard wrote: The price of RealSpeak is not far from an order of magnitude higher compared to Cepstral. Only an order of magnitude? They've reduced it a lot then. :-) Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
Erik - Have you found RealSpeak to be worth the cost? Actually my last note was probably a bit misleading because in the particular cases I mentioned RealSpeak, the platform wasn't Asterisk and Cepstral wasn't even on the radar. Can Cepstral, with the hourly $ spent on tuning, be made to be a reasonable substitute? Nuance would say no :) I'd say maybe. Call up +14164854854, it's a recent project we did for a client using Asterisk, Cepstral, and a lot of custom code. It's a free phone-in service that allows folks to get local traffic, weather, news, commuter transit, border crossing wait times, and more. There's obviously quite a bit of domain-specific, dynamic, constantly changing text, so this is certainly an example of pushing it to the max. Just think of all the street names it has the potential to mispronounce. It's a work in progress, but it's very promising. Definitely an example of a lot of hourly $ spent on tuning as you put it. My results: The RealSpeak sample was more clear than the Cepstral. Depends on what you mean by more clear. As Brent Davidson mentions, make sure you're comparing 8khz to 8khz, or similar. If you mean it pronounces things better, then I agree. That being said, I'd really be interested in hearing if anyone has done a RealSpeak-to-Asterisk conduit. I wasn't able to quickly uncover how they interact with third-party systems - is it VoIP? A C library? Some sort of HTTP socket? The more methods we can get working with Asterisk, the better, because not every implementation of a voice system has the same requirements... MRCP is the standard for interfacing with ASR and TTS engines (including RealSpeak) in other platforms. Brief Googling reveals a previous flame war on asterisk-dev regarding MRCP. I have no idea if it's implemented in Asterisk now. Regards, -- Erik Caneris Tel: 647-723-6365 Fax: 647-723-5365 Toll-free: 1-866-827-0021 www.caneris.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral vs festival
2008/12/3 Steve Underwood [EMAIL PROTECTED] Jean-Denis Girard wrote: The price of RealSpeak is not far from an order of magnitude higher compared to Cepstral. Only an order of magnitude? They've reduced it a lot then. :-) 1 order of magnitude = x10 Then, shall we say 500$/simultaneous voice ? Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cepstral ... Swift... weird result
Asterisk 1.2, and Cepstral 5, Allison voice. I execute: swift Please enter your pin. -o please-enter-your-pin.ulaw -p audio/channels=1,audio/encoding=ulaw,audio/sampling-rate=8000 then copy it up to /var/lib/asterisk/sounds, and Play() the file. The sound file seems corrupted. All I hear is 'please' or 'please' followed by the rest of the sentence said so fast I almost can't hear it. I've tried other various of the -p option to swift, same results. Also tried generating a wav file and converting to ulaw with sox, same result. I did this once before and it worked. What am I doing wrong? Doug. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral TTS and app_swift
what versions of asterisk on both systems ? On 6/5/07, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: Have you tried something along the lines of: System(swift blah blah blah -o blah.wav) Playback(blah.wav) It does have an inherent delay for the generation step but maybe swift binary segfaults less? I've only used cepstral via swift binary, and it has never segfaulted for me. My swift and voice are version 4.2.0. I doubt different voices behave differently, but just in case, I use the $7 Damien voice. Moj Julian Lyndon-Smith wrote: We are having some major problems with app_swift since we went live. It is regularly segfaulting. I don't know if this is my fault or not, but here's the story: Installed the cepstral voices (at the time, 4.0) on our test system (2.6.9-42.0.10.ELsmp) and later added some extra voices (now 4.2). All worked fine - we stress tested (20+ simultaneous calls). Move to live ( 2.6.9-22.0.1.ELsmp) . Installed the cepstral voices (only 4.2). Started having problems with only 5 calls: swift by itself on the command line was fine (it worked) but app_swift complained that it couldn't find any voices. Looking into /opt/swift/lib, I saw that it was different to my test system. On live I had (snipped) -rwxrwxrwx 1 root root 139612 Jun 1 23:10 libceplang_en.so =rwxrwxrwx 1 root root 139612 Jun 1 23:11 libceplang_en.so.4 -rwxr-xr-x 1 root root 139612 Jun 1 07:09 libceplang_en.so.4.2 -rwxrwxrwx 1 root root 547624 Jun 1 23:11 libceplex_uk.so -rwxrwxrwx 1 root root 547624 Jun 1 23:11 libceplex_uk.so.4 -rwxr-xr-x 1 root root 547624 Jun 1 07:09 libceplex_uk.so.4.2 on test I had lrwxrwxrwx 1 999 20202 20 Apr 24 16:17 libceplang_en.so - libceplang_en.so.4.2 lrwxrwxrwx 1 999 20202 20 Apr 24 16:17 libceplang_en.so.4 - libceplang_en.so.4.2 -rwxrwxr-x 1 999 20202 315933 Aug 17 2006 libceplang_en.so.4.1 -rwxrwxr-x 1 999 20202 139612 Mar 15 18:21 libceplang_en.so.4.2 lrwxrwxrwx 1 999 20202 19 Apr 24 16:17 libceplex_uk.so - libceplex_uk.so.4.2 lrwxrwxrwx 1 999 20202 19 Apr 24 16:17 libceplex_uk.so.4 - libceplex_uk.so.4.2 -rwxrwxr-x 1 999 20202 591033 Aug 17 2006 libceplex_uk.so.4.1 -rwxrwxr-x 1 999 20202 547624 Mar 15 18:20 libceplex_uk.so.4.2 I then removed all the non 4.2 libs and created a symbolic link to the 4.2 libs to match test. fired it all up, and app_swift then worked. Or so I thought. segfault - but not on every call. what I would like to know is: A) has anybody got a later version of app_swift (0.9.1) B) does anyone else use cepstral, and how ? C) what is the story with the cepstral libraries ? many thanks Julian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral TTS and app_swift
Thanks for the input - that's what we've ended up doing. I was concerned at the impact on system performance, but it seems negligible. I tested it with 30 simultaneous calls (1 calls in total) using sipp and it didn't crash once. Julian Mojo with Horan Company, LLC wrote: Have you tried something along the lines of: System(swift blah blah blah -o blah.wav) Playback(blah.wav) It does have an inherent delay for the generation step but maybe swift binary segfaults less? I've only used cepstral via swift binary, and it has never segfaulted for me. My swift and voice are version 4.2.0. I doubt different voices behave differently, but just in case, I use the $7 Damien voice. Moj Julian Lyndon-Smith wrote: We are having some major problems with app_swift since we went live. It is regularly segfaulting. I don't know if this is my fault or not, but here's the story: Installed the cepstral voices (at the time, 4.0) on our test system (2.6.9-42.0.10.ELsmp) and later added some extra voices (now 4.2). All worked fine - we stress tested (20+ simultaneous calls). Move to live ( 2.6.9-22.0.1.ELsmp) . Installed the cepstral voices (only 4.2). Started having problems with only 5 calls: swift by itself on the command line was fine (it worked) but app_swift complained that it couldn't find any voices. Looking into /opt/swift/lib, I saw that it was different to my test system. On live I had (snipped) -rwxrwxrwx 1 root root 139612 Jun 1 23:10 libceplang_en.so =rwxrwxrwx 1 root root 139612 Jun 1 23:11 libceplang_en.so.4 -rwxr-xr-x 1 root root 139612 Jun 1 07:09 libceplang_en.so.4.2 -rwxrwxrwx 1 root root 547624 Jun 1 23:11 libceplex_uk.so -rwxrwxrwx 1 root root 547624 Jun 1 23:11 libceplex_uk.so.4 -rwxr-xr-x 1 root root 547624 Jun 1 07:09 libceplex_uk.so.4.2 on test I had lrwxrwxrwx 1 999 20202 20 Apr 24 16:17 libceplang_en.so - libceplang_en.so.4.2 lrwxrwxrwx 1 999 20202 20 Apr 24 16:17 libceplang_en.so.4 - libceplang_en.so.4.2 -rwxrwxr-x 1 999 20202 315933 Aug 17 2006 libceplang_en.so.4.1 -rwxrwxr-x 1 999 20202 139612 Mar 15 18:21 libceplang_en.so.4.2 lrwxrwxrwx 1 999 20202 19 Apr 24 16:17 libceplex_uk.so - libceplex_uk.so.4.2 lrwxrwxrwx 1 999 20202 19 Apr 24 16:17 libceplex_uk.so.4 - libceplex_uk.so.4.2 -rwxrwxr-x 1 999 20202 591033 Aug 17 2006 libceplex_uk.so.4.1 -rwxrwxr-x 1 999 20202 547624 Mar 15 18:20 libceplex_uk.so.4.2 I then removed all the non 4.2 libs and created a symbolic link to the 4.2 libs to match test. fired it all up, and app_swift then worked. Or so I thought. segfault - but not on every call. what I would like to know is: A) has anybody got a later version of app_swift (0.9.1) B) does anyone else use cepstral, and how ? C) what is the story with the cepstral libraries ? many thanks Julian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email for dotr.com has been scanned by MessageLabs __ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cepstral TTS and app_swift
We are having some major problems with app_swift since we went live. It is regularly segfaulting. I don't know if this is my fault or not, but here's the story: Installed the cepstral voices (at the time, 4.0) on our test system (2.6.9-42.0.10.ELsmp) and later added some extra voices (now 4.2). All worked fine - we stress tested (20+ simultaneous calls). Move to live ( 2.6.9-22.0.1.ELsmp) . Installed the cepstral voices (only 4.2). Started having problems with only 5 calls: swift by itself on the command line was fine (it worked) but app_swift complained that it couldn't find any voices. Looking into /opt/swift/lib, I saw that it was different to my test system. On live I had (snipped) -rwxrwxrwx 1 root root 139612 Jun 1 23:10 libceplang_en.so =rwxrwxrwx 1 root root 139612 Jun 1 23:11 libceplang_en.so.4 -rwxr-xr-x 1 root root 139612 Jun 1 07:09 libceplang_en.so.4.2 -rwxrwxrwx 1 root root 547624 Jun 1 23:11 libceplex_uk.so -rwxrwxrwx 1 root root 547624 Jun 1 23:11 libceplex_uk.so.4 -rwxr-xr-x 1 root root 547624 Jun 1 07:09 libceplex_uk.so.4.2 on test I had lrwxrwxrwx 1 999 20202 20 Apr 24 16:17 libceplang_en.so - libceplang_en.so.4.2 lrwxrwxrwx 1 999 20202 20 Apr 24 16:17 libceplang_en.so.4 - libceplang_en.so.4.2 -rwxrwxr-x 1 999 20202 315933 Aug 17 2006 libceplang_en.so.4.1 -rwxrwxr-x 1 999 20202 139612 Mar 15 18:21 libceplang_en.so.4.2 lrwxrwxrwx 1 999 20202 19 Apr 24 16:17 libceplex_uk.so - libceplex_uk.so.4.2 lrwxrwxrwx 1 999 20202 19 Apr 24 16:17 libceplex_uk.so.4 - libceplex_uk.so.4.2 -rwxrwxr-x 1 999 20202 591033 Aug 17 2006 libceplex_uk.so.4.1 -rwxrwxr-x 1 999 20202 547624 Mar 15 18:20 libceplex_uk.so.4.2 I then removed all the non 4.2 libs and created a symbolic link to the 4.2 libs to match test. fired it all up, and app_swift then worked. Or so I thought. segfault - but not on every call. what I would like to know is: A) has anybody got a later version of app_swift (0.9.1) B) does anyone else use cepstral, and how ? C) what is the story with the cepstral libraries ? many thanks Julian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral TTS and app_swift
Have you tried something along the lines of: System(swift blah blah blah -o blah.wav) Playback(blah.wav) It does have an inherent delay for the generation step but maybe swift binary segfaults less? I've only used cepstral via swift binary, and it has never segfaulted for me. My swift and voice are version 4.2.0. I doubt different voices behave differently, but just in case, I use the $7 Damien voice. Moj Julian Lyndon-Smith wrote: We are having some major problems with app_swift since we went live. It is regularly segfaulting. I don't know if this is my fault or not, but here's the story: Installed the cepstral voices (at the time, 4.0) on our test system (2.6.9-42.0.10.ELsmp) and later added some extra voices (now 4.2). All worked fine - we stress tested (20+ simultaneous calls). Move to live ( 2.6.9-22.0.1.ELsmp) . Installed the cepstral voices (only 4.2). Started having problems with only 5 calls: swift by itself on the command line was fine (it worked) but app_swift complained that it couldn't find any voices. Looking into /opt/swift/lib, I saw that it was different to my test system. On live I had (snipped) -rwxrwxrwx 1 root root 139612 Jun 1 23:10 libceplang_en.so =rwxrwxrwx 1 root root 139612 Jun 1 23:11 libceplang_en.so.4 -rwxr-xr-x 1 root root 139612 Jun 1 07:09 libceplang_en.so.4.2 -rwxrwxrwx 1 root root 547624 Jun 1 23:11 libceplex_uk.so -rwxrwxrwx 1 root root 547624 Jun 1 23:11 libceplex_uk.so.4 -rwxr-xr-x 1 root root 547624 Jun 1 07:09 libceplex_uk.so.4.2 on test I had lrwxrwxrwx 1 999 20202 20 Apr 24 16:17 libceplang_en.so - libceplang_en.so.4.2 lrwxrwxrwx 1 999 20202 20 Apr 24 16:17 libceplang_en.so.4 - libceplang_en.so.4.2 -rwxrwxr-x 1 999 20202 315933 Aug 17 2006 libceplang_en.so.4.1 -rwxrwxr-x 1 999 20202 139612 Mar 15 18:21 libceplang_en.so.4.2 lrwxrwxrwx 1 999 20202 19 Apr 24 16:17 libceplex_uk.so - libceplex_uk.so.4.2 lrwxrwxrwx 1 999 20202 19 Apr 24 16:17 libceplex_uk.so.4 - libceplex_uk.so.4.2 -rwxrwxr-x 1 999 20202 591033 Aug 17 2006 libceplex_uk.so.4.1 -rwxrwxr-x 1 999 20202 547624 Mar 15 18:20 libceplex_uk.so.4.2 I then removed all the non 4.2 libs and created a symbolic link to the 4.2 libs to match test. fired it all up, and app_swift then worked. Or so I thought. segfault - but not on every call. what I would like to know is: A) has anybody got a later version of app_swift (0.9.1) B) does anyone else use cepstral, and how ? C) what is the story with the cepstral libraries ? many thanks Julian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral voices
Kai-Uwe Jensen wrote: There's also an app_swift available at http://www.loopfree.net/app_swift/ Thanks to all that responded. I've used app_swift as mentioned above and it suits my needs. Thanks again Julian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cepstral and numbers
Does anyone have any idea on how to force cepstral to convert a number to speech ? I have noticed that sometimes it speaks the number correctly, and at others it doesn't. 1) 787 is pronounced 7-8-7 2) 123 is pronounced one-hundred and twenty-three. 1) is wrong for what i need, 2) is perfect. Is there anyway of forcing numbers to be pronounced as 2) ? I've tried looking at the ssml tags .. TIA Julian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral and numbers
Oh man - the second I send this, I find the answer. say-as type=currency12345.44/say-as Sorry for the waste of bandwidth. Julian Julian Lyndon-Smith wrote: Does anyone have any idea on how to force cepstral to convert a number to speech ? I have noticed that sometimes it speaks the number correctly, and at others it doesn't. 1) 787 is pronounced 7-8-7 2) 123 is pronounced one-hundred and twenty-three. 1) is wrong for what i need, 2) is perfect. Is there anyway of forcing numbers to be pronounced as 2) ? I've tried looking at the ssml tags .. TIA Julian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ __ This email for dotr.com has been scanned by MessageLabs __ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral and numbers
Julian Lyndon-Smith wrote: Does anyone have any idea on how to force cepstral to convert a number to speech ? I have noticed that sometimes it speaks the number correctly, and at others it doesn't. 1) 787 is pronounced 7-8-7 2) 123 is pronounced one-hundred and twenty-three. 1) is wrong for what i need, 2) is perfect. Is there anyway of forcing numbers to be pronounced as 2) ? I've tried looking at the ssml tags .. TIA Julian Total guess on my part, but by any chance does it use 1) if the number isn't separated by whitespace from something preceeding it, but 2) if it is? This would make a traffic report for example read correctly: The traffic on I-84 is moving at 60 MPH. Where you'd want I-84 read using 1) and 60 using 2) Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral and numbers
Julian Lyndon-Smith wrote:
Does anyone have any idea on how to force cepstral to convert a number
to speech ?
I have noticed that sometimes it speaks the number correctly, and at
others it doesn't.
1) 787 is pronounced 7-8-7
2) 123 is pronounced one-hundred and twenty-three.
You could probably place dots between the numbers if you want to say
each number individually.
Swift.PlayString('1.2.3.4.', false);
--
Warm Regards,
Lee
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Re: [asterisk-users] Cepstral voices
Lee Jenkins wrote:
Funny you should mention FastAGI. I am implementing a variation of my
DTSwift app through an Object Pascal based FastAGI scripting server now.
http://www.datatrakpos.com/pos/datatalk/images/asterpas.htm
The newer version just uses the System() AGI command to build the file
to play through the shell. I'd be open to any suggestions for a more
efficient way of doing it.
Sorry for not answering faster - was busy shoveling snow.
Once you've got app_swift installed, saying something
should be pretty close to:
AGI.Exec('Swift','This is something to say.');
And since it doesn't render to a file first you'll probably experience
less of a delay to say something.
Steve
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Re: [asterisk-users] Cepstral voices
Best code comment ever, by the way: here's a for loop for i := 1 to iLen do :-) On 3/16/07, Lee Jenkins [EMAIL PROTECTED] wrote: Steve Prior wrote: Julian Lyndon-Smith wrote: what input text ? To what application ? I agree completely with the app_swift suggestion from loopfree as Kai suggested. It provides the app_Swift which you can use from within a dialplan. In fact, if you're getting fancy by using a fastAGI bound language(as I'm doing with asterisk-java), app_swift becomes the only good option. slight rantI think Cepstral should be providing an app_swift like binding themselves because if you're writing an application which is going to use information from a back end business model in creating the speech (and this is something they seem to think is their future and I agree), then a high level language through fastAGI seems by far the best way to control the call. /slight rant Funny you should mention FastAGI. I am implementing a variation of my DTSwift app through an Object Pascal based FastAGI scripting server now. http://www.datatrakpos.com/pos/datatalk/images/asterpas.htm The newer version just uses the System() AGI command to build the file to play through the shell. I'd be open to any suggestions for a more efficient way of doing it. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- sean ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral voices
Sean Bright wrote: Best code comment ever, by the way: here's a for loop for i := 1 to iLen do :-) LOL, I know. The script was originally to show some of the standard language features supported by the scripting engine. But then I started writing all the db access and cepstral examples and got off on a bit of a tangent... -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral voices
Steve Prior wrote:
Lee Jenkins wrote:
Funny you should mention FastAGI. I am implementing a variation of my
DTSwift app through an Object Pascal based FastAGI scripting server now.
http://www.datatrakpos.com/pos/datatalk/images/asterpas.htm
The newer version just uses the System() AGI command to build the file
to play through the shell. I'd be open to any suggestions for a more
efficient way of doing it.
Sorry for not answering faster - was busy shoveling snow.
Once you've got app_swift installed, saying something
should be pretty close to:
AGI.Exec('Swift','This is something to say.');
And since it doesn't render to a file first you'll probably experience
less of a delay to say something.
Yeah, that is what mine does. I'll take a look at loopfree code and see
how it's streamed.
--
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Lee
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[asterisk-users] Cepstral voices
what's the easiest way of using cepstral voices with asterisk ? On their website, in the ssml page (http://www.cepstral.com/cgi-bin/support?page=ssml), they say Asterisk PBX SSML can be used with Cepstral voices in Asterisk by simply embedding the markup into the input text. what input text ? To what application ? Thanks ! Julian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral voices
Julian Lyndon-Smith wrote: what's the easiest way of using cepstral voices with asterisk ? On their website, in the ssml page (http://www.cepstral.com/cgi-bin/support?page=ssml), they say Asterisk PBX SSML can be used with Cepstral voices in Asterisk by simply embedding the markup into the input text. what input text ? To what application ? I wrote an AGI wrapper for Cepstral a while back. It's a freepascal application. There is a binary included along with pascal source: http://www.voip-info.org/wiki/index.php?page=DTSwift%20Cepstral%20AGI%20Wrapper There are also some good scripts available like this one: http://www.voip-info.org/wiki/view/swift.agi Or you can use the System() application to call the relevant shell commands for Capstal Swift application: http://www.voip-info.org/wiki/view/Swift HIH -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral voices
There's also an app_swift available at http://www.loopfree.net/app_swift/ -- I am Dyslexic of Borg. Fusistance is retile. Your ass will be laminated! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral voices
Julian Lyndon-Smith wrote: what's the easiest way of using cepstral voices with asterisk ? On their website, in the ssml page (http://www.cepstral.com/cgi-bin/support?page=ssml), they say Asterisk PBX SSML can be used with Cepstral voices in Asterisk by simply embedding the markup into the input text. what input text ? To what application ? I agree completely with the app_swift suggestion from loopfree as Kai suggested. It provides the app_Swift which you can use from within a dialplan. In fact, if you're getting fancy by using a fastAGI bound language(as I'm doing with asterisk-java), app_swift becomes the only good option. slight rantI think Cepstral should be providing an app_swift like binding themselves because if you're writing an application which is going to use information from a back end business model in creating the speech (and this is something they seem to think is their future and I agree), then a high level language through fastAGI seems by far the best way to control the call. /slight rant Stay away from app_cepstral, if you're Googling it appears to be an option, but it didn't work for me. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral voices
Steve Prior wrote: Julian Lyndon-Smith wrote: what input text ? To what application ? I agree completely with the app_swift suggestion from loopfree as Kai suggested. It provides the app_Swift which you can use from within a dialplan. In fact, if you're getting fancy by using a fastAGI bound language(as I'm doing with asterisk-java), app_swift becomes the only good option. slight rantI think Cepstral should be providing an app_swift like binding themselves because if you're writing an application which is going to use information from a back end business model in creating the speech (and this is something they seem to think is their future and I agree), then a high level language through fastAGI seems by far the best way to control the call. /slight rant Funny you should mention FastAGI. I am implementing a variation of my DTSwift app through an Object Pascal based FastAGI scripting server now. http://www.datatrakpos.com/pos/datatalk/images/asterpas.htm The newer version just uses the System() AGI command to build the file to play through the shell. I'd be open to any suggestions for a more efficient way of doing it. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral voice still nags after registration
Thanks Paul. I think it was nagging because the phpagi code looks to see if there is already a wav file before creating a new one. Since I had old ones with the nagging, it didn't create new ones. The problem I am having now is that it won't play it at all, just beeps. Thanks! On 1/12/07, Paul [EMAIL PROTECTED] wrote: blackwater dev wrote: I'm using trixbox and the asterisk agi. I downloaded a cepstral voice and worked with it until I got the code to do what I wanted. I then registered the voice today to get rid of the 'this voice is not yet registered, stuff yet it still does that. Any ideas on how to fix this? It told me my info was valid. Thanks! I am not using trixbox and I installed swift in /opt In my case the file of interest is: /opt/swift/voices/Diane/license.txt The file contains my name, company and a license key See if you have a license.txt file like that in the right place Set it to root:root ownership and 644 permissions ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cepstral voice still nags after registration
I'm using trixbox and the asterisk agi. I downloaded a cepstral voice and worked with it until I got the code to do what I wanted. I then registered the voice today to get rid of the 'this voice is not yet registered, stuff yet it still does that. Any ideas on how to fix this? It told me my info was valid. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cepstral voice still nags after registration
blackwater dev wrote: I'm using trixbox and the asterisk agi. I downloaded a cepstral voice and worked with it until I got the code to do what I wanted. I then registered the voice today to get rid of the 'this voice is not yet registered, stuff yet it still does that. Any ideas on how to fix this? It told me my info was valid. Thanks! I am not using trixbox and I installed swift in /opt In my case the file of interest is: /opt/swift/voices/Diane/license.txt The file contains my name, company and a license key See if you have a license.txt file like that in the right place Set it to root:root ownership and 644 permissions ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cepstral/Swift TTS app
Hey everyone, I was frustrated with the existing app_cepstral/app_swift TTS modules I've found on the net, so I hacked up my own. It's been working really well for me so I thought I'd share. In developing this, I wanted to avoid: * the startup delay incurred writing TTS output to a temp file then playing it back * gluing cepstral/swift to the asterisk module in a way that fubard the cepstral licensing server concurrency count And I wanted to support: * configurable in-memory buffering so you can balance how far ahead swift will get in generating audio. So you can trade off swift process concurrency vs memory use. * hanging up or exiting the speech call with DTMF is handled quickly and gracefully download and more comments at: http://www.loopfree.net/app_swift/ If anyone can make use of this, great! Or let me know if it's horribly broken. All I can say at this point is that it works really well for me in a TTS/IVR system I'm developing. -- -Will Orton :: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cepstral and Asterisk again...
In the show app cepstral...it gives an example of if you have more than 1 voice... exten= 1,1,Cepstral(voice name="William"hello world/voice) However that doesn't work...it will still complain it can't find the voice...etc...etc... Yet if you have only one voice in the directory it is fine... Anyone run into this problem? Thanks, Don ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cepstral and Asterisk
Has anyone used Cepstral for text to speech before? I am testing the demo and it seems to take about 20 seconds for the speech to start... On a 3.4Ghz 2GB machine... Thanks, Don ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral and Asterisk
On Wednesday August 16 2006 7:01 pm, Don wrote: Has anyone used Cepstral for text to speech before? I am testing the demo and it seems to take about 20 seconds for the speech to start... On a 3.4Ghz 2GB machine... Thanks, Don Don, I have been using Cepstral for about a year now and it has worked very well. It starts speaking almost immediately. You definitely know it is a computer voice but thats okay for my application. The following is swift.agi that I found on voip-info.org (I cant remember the author or I would credit him here, my apologies.) The last line documents how to use. ## #!/bin/sh #Assign the value sent from the exten= line to $text so it can be used below text=`echo $*` #Set $stdin to something stdin=0 while [ $stdin != ] do read stdin if [ $stdin != ] then stdin2=`echo $stdin | sed -e 's/: /=/' -e 's///g' -e 's/$//' -e 's/=/=/'` eval `echo $stdin2` fi done calleridnum=`echo $agi_callerid | cut -f2 -d\ | cut -f1 -d\` calleridname=`echo $agi_callerid | cut -f1 -d\ ` /opt/swift/bin/swift -o /tmp/$agi_uniqueid.wav -p audio/channels=1,audio/sampling-rate=8000 $text #Now, tell asterisk to play that file echo stream file /tmp/$agi_uniqueid # #Read the reply from asterisk to our command read stream #Clean up our mess and delete that file rm /tmp/$agi_uniqueid.wav exit 0 # exten= s,1,agi(swift.agi|This is some text\, which needs to be converted to speech.) ## I have used this (on a very low call volume obviously) on as low end a machine as PII 400 with 512 meg ram. Hope this helps -- John Millican Senior Partner Director of Technology Sentinel Communications PO Box 9 Wentworth, NH 03282 (603) 764-9163 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cepstral and Asterisk
Ahh...this is what I thought I might have to do...also...
It seems they are outputting to a tmp file with cepstral and then playing
that file...instead of actually saying it on the fly in the dialplan.
If I do Cepstral(${foo}) it takes a damn long time to start heh...too long
for any telecom app...
- Original Message -
From: John Millican [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, August 16, 2006 9:10 PM
Subject: Re: [asterisk-users] Cepstral and Asterisk
On Wednesday August 16 2006 7:01 pm, Don wrote:
Has anyone used Cepstral for text to speech before? I am testing the demo
and it seems to take about 20 seconds for the speech to start... On a
3.4Ghz 2GB machine...
Thanks,
Don
Don,
I have been using Cepstral for about a year now and it has worked very
well.
It starts speaking almost immediately. You definitely know it is a
computer
voice but thats okay for my application.
The following is swift.agi that I found on voip-info.org (I cant remember
the
author or I would credit him here, my apologies.) The last line documents
how to use.
##
#!/bin/sh
#Assign the value sent from the exten= line to $text so it can be used
below
text=`echo $*`
#Set $stdin to something
stdin=0
while [ $stdin != ]
do
read stdin
if [ $stdin != ]
then
stdin2=`echo $stdin | sed -e 's/: /=/' -e 's///g' -e 's/$//' -e
's/=/=/'`
eval `echo $stdin2`
fi
done
calleridnum=`echo $agi_callerid | cut -f2 -d\ | cut -f1 -d\`
calleridname=`echo $agi_callerid | cut -f1 -d\ `
/opt/swift/bin/swift -o /tmp/$agi_uniqueid.wav -p
audio/channels=1,audio/sampling-rate=8000 $text
#Now, tell asterisk to play that file
echo stream file /tmp/$agi_uniqueid #
#Read the reply from asterisk to our command
read stream
#Clean up our mess and delete that file
rm /tmp/$agi_uniqueid.wav
exit 0
# exten= s,1,agi(swift.agi|This is some text\, which needs to be
converted to speech.)
##
I have used this (on a very low call volume obviously) on as low end a
machine
as PII 400 with 512 meg ram.
Hope this helps
--
John Millican
Senior Partner
Director of Technology
Sentinel Communications
PO Box 9
Wentworth, NH 03282
(603) 764-9163
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Checked by AVG Free Edition.
Version: 7.1.405 / Virus Database: 268.11.0/420 - Release Date: 8/16/2006
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Re: [asterisk-users] Cepstral and Asterisk
Of course...doing it this way also...kind of negates the need for app_cepstral.so being loaded...can just use Playback or background...whatever... Thanks again though...verified my idea on what I would need to do. - Original Message - From: John Millican [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, August 16, 2006 9:10 PM Subject: Re: [asterisk-users] Cepstral and Asterisk On Wednesday August 16 2006 7:01 pm, Don wrote: Has anyone used Cepstral for text to speech before? I am testing the demo and it seems to take about 20 seconds for the speech to start... On a 3.4Ghz 2GB machine... Thanks, Don Don, I have been using Cepstral for about a year now and it has worked very well. It starts speaking almost immediately. You definitely know it is a computer voice but thats okay for my application. The following is swift.agi that I found on voip-info.org (I cant remember the author or I would credit him here, my apologies.) The last line documents how to use. ## #!/bin/sh #Assign the value sent from the exten= line to $text so it can be used below text=`echo $*` #Set $stdin to something stdin=0 while [ $stdin != ] do read stdin if [ $stdin != ] then stdin2=`echo $stdin | sed -e 's/: /=/' -e 's///g' -e 's/$//' -e 's/=/=/'` eval `echo $stdin2` fi done calleridnum=`echo $agi_callerid | cut -f2 -d\ | cut -f1 -d\` calleridname=`echo $agi_callerid | cut -f1 -d\ ` /opt/swift/bin/swift -o /tmp/$agi_uniqueid.wav -p audio/channels=1,audio/sampling-rate=8000 $text #Now, tell asterisk to play that file echo stream file /tmp/$agi_uniqueid # #Read the reply from asterisk to our command read stream #Clean up our mess and delete that file rm /tmp/$agi_uniqueid.wav exit 0 # exten= s,1,agi(swift.agi|This is some text\, which needs to be converted to speech.) ## I have used this (on a very low call volume obviously) on as low end a machine as PII 400 with 512 meg ram. Hope this helps -- John Millican Senior Partner Director of Technology Sentinel Communications PO Box 9 Wentworth, NH 03282 (603) 764-9163 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.405 / Virus Database: 268.11.0/420 - Release Date: 8/16/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cepstral , options to read the contents of a file
Hi I had installed Cepstral , and it is working in Asterisk , it workfine for exten = s,1,Answer exten = s,2,Wait(1) exten = s,3,Cepstral( This is Just a test ) exten = s,4,Cepstral(Hope u are getting this voices) but instead of the text contents for Cepstral , can I use the file name location , where it can read the file Thanks Joseph John ___ 24 FIFA World Cup tickets to be won with Yahoo! Mail http://uk.mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cepstral , options to read the contents of a file
Hi, You can call an agi script to convert the text file to wave format. Example: http://www.voip-info.org/wiki/view/swift.agi Kevin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Joseph Sent: Monday, May 01, 2006 7:08 PM To: Asterisk Users Subject: [Asterisk-Users] Cepstral , options to read the contents of a file Hi I had installed Cepstral , and it is working in Asterisk , it workfine for exten = s,1,Answer exten = s,2,Wait(1) exten = s,3,Cepstral( This is Just a test ) exten = s,4,Cepstral(Hope u are getting this voices) but instead of the text contents for Cepstral , can I use the file name location , where it can read the file Thanks Joseph John ___ 24 FIFA World Cup tickets to be won with Yahoo! Mail http://uk.mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cepstral in AGI problem
I'm expirimenting with Cepstral via swift.agi. It loads w/ no problem, but there is a fairly long separation between words -- almost as if it is processing one word at a time, rather than stringing a phrase together. When I run cepstral with a similar script, but not through an actual call from asterisk the separation between words is short and natural.I'd really appreciate any help. Thanks Yahoo! Photos Got holiday prints? See all the ways to get quality prints in your hands ASAP.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral in AGI problem
Hi Wolfgang
I've had the same issue.
I don't think agi per se is the problem but rather the way you are
calling the swift excutable.
i.e.
swift this is a test
gets rewritten as
swift this is a test
Hope this makes sense.
FYI, here's how I do it in fastagi/java:
--
String toSay = some text goes here for David to say;
String[] args = {/usr/bin/swift, -n, David, toSay, -o,
/tmp/test.wav};
Runtime.getRuntime().exec(args).waitFor();
--
BR,
Cristi
On 1/13/06, Wolfgang Borgon [EMAIL PROTECTED] wrote:
I'm expirimenting with Cepstral via swift.agi. It loads w/ no problem, but
there is a fairly long separation between words -- almost as if it is
processing one word at a time, rather than stringing a phrase together.
When I run cepstral with a similar script, but not through an actual call
from asterisk the separation between words is short and natural.
I'd really appreciate any help.
Thanks
Yahoo! Photos
Got holiday prints? See all the ways to get quality prints in your hands
ASAP.
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Re: [Asterisk-Users] Cepstral
On Monday 11 Jul 2005 05:02, Michael Stearne wrote: On 7/10/05, Jim Archer [EMAIL PROTECTED] wrote: Thanks William and John, I'll look again for that download. Comments below... --On Sunday, July 10, 2005 1:50 PM +0200 Wilson Pickett [EMAIL PROTECTED] wrote: FWIW? I bought that voice and I find it amusing, but not ready for prime time. I had it read articles from a publication and it was ludicrous. I can understand the people talking about ATT, I think I heard a demo that was very convincing. What is ATT? Is it another text to speech engine? I installed Festival a ATT Natural voices seem to be pretty good. You can hear samples here: http://www.wizzardsoftware.com/att_desktop.php . The Rich voice I think sounds the best. They are a little better than the Cepstral voices. But the Cepstral voices are vey good also. Compared to Rhetorical (http://www.rhetorical.com/cgi-bin/demo.cgi), the ATT system sounds awful. B ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral
On Mon, 2005-07-11 at 09:19 +0100, Bob Goddard wrote: Compared to Rhetorical (http://www.rhetorical.com/cgi-bin/demo.cgi), the ATT system sounds awful. Yes it does sound considerably better, but what do I know I have a hearing loss. Anyway, have you managed to integrate this with asterisk successfully? If so how? use the generated output (wav presumably) and stream it via an AGI? Or does it have more direct asterisk connectivity? I had written a script that would snarf sitepal TTS data, rip the SWF to get the resulting mp3, transcode that to something more appropriate, and cache it for future requests of the same data. I called this script via a macro so it was trivial to specify a voice and text to speak in a dialplan entry. The whole process was quite trivial but not as system friendly as I like (too many fork and execs for my taste). Plus becuase the TTS engine runs on their systems there is noticable lag in actually getting the data - part of which appears to be their system is somewhat slow in generating the audio data, but in this instance free. I could of course do something similar with Rhetorical, even if its running on my own machine, but I just dont like doing it that way, it works, but seems wrong due to many wasted cpu cycles. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral
Compared to Rhetorical (http://www.rhetorical.com/cgi-bin/demo.cgi), the ATT system sounds awful. You're 100% correct! My mistake, I was thinking of rhetorical when I said ATT. I'm not familiar with ATT at all - my bad! Thanks for correcting this and reminding me of rhetorical. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cepstral
I have been reading about Cepstral, their voices and the Digium partner agreement with them. I see where they sell the voices and the licenses for them, but what I can't find is how to buy or get Swift? If I understand correctly, swift is the actual program that makes the speech? Strangely, the Cepstral web site does not explain this... Can someone shed some light? Thanks... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral
what I can't find is how to buy or get Swift? If I understand correctly, swift is the actual program that makes the speech? IIRC, you can download everything you need to make the thing talk, including a voice like David. It works exactly like it will when you buy a license except there is some kind of crippling until you install the license key. I don't remember if this is a statement made by the voice each time or a time out. FWIW? I bought that voice and I find it amusing, but not ready for prime time. I had it read articles from a publication and it was ludicrous. I can understand the people talking about ATT, I think I heard a demo that was very convincing. So much depends on what you are trying to do. I just wanted to have a way to allow asterisk to talk in a demo, to show the concept. Unfortunately, showing a talking server with Cepestral's David is little like showing a prototype website: people don't always have the imagination (like we all do here :) to see what this would be like when actually done (or using a better voice in this case). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral
what I can't find is how to buy or get Swift? If I understand correctly, swift is the actual program that makes the speech? IIRC, you can download everything you need to make the thing talk, including a voice like David. It works exactly like it will when you buy a license except there is some kind of crippling until you install the license key. I don't remember if this is a statement made by the voice each time or a time out. FWIW? I bought that voice and I find it amusing, but not ready for prime time. I had it read articles from a publication and it was ludicrous. I can understand the people talking about ATT, I think I heard a demo that was very convincing. So much depends on what you are trying to do. I just wanted to have a way to allow asterisk to talk in a demo, to show the concept. Unfortunately, showing a talking server with Cepestral's David is little like showing a prototype website: people don't always have the imagination (like we all do here :) to see what this would be like when actually done (or using a better voice in this case). I have the emily voice and she sounds much like the marine weather station reports. the crippling is just a message that says it is an unregistered version or the like. Yes you can absolutely tell that it is speech synthesis but it is understandable. You can fiddle with the settings, in the readme this is explained, and make it sound a little better. John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral
I have been reading about Cepstral, their voices and the Digium partner agreement with them. I see where they sell the voices and the licenses for them, but what I can't find is how to buy or get Swift? If I understand correctly, swift is the actual program that makes the speech? Strangely, the Cepstral web site does not explain this... Can someone shed some light? Thanks... I have been using cepstral for a while now. Swift is the old name(I believe) for cepstral and is placed in the /install_dir/bin directory when you unpack the cepstral download. John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral
Thanks William and John, I'll look again for that download. Comments below... --On Sunday, July 10, 2005 1:50 PM +0200 Wilson Pickett [EMAIL PROTECTED] wrote: FWIW? I bought that voice and I find it amusing, but not ready for prime time. I had it read articles from a publication and it was ludicrous. I can understand the people talking about ATT, I think I heard a demo that was very convincing. What is ATT? Is it another text to speech engine? I installed Festival a few days ago and have been playing with it. It sounds okay, but I decided to look to see if I could find something better. Some searching on this list and elsewhere revealed that people were raving about Cepstral, so I figured I would try it. I found their demo page and, honestly, didn't think it sounded much better than Festival. But I like that it had different voice options and Festival seems to have an Irish accent. Not that I mind an Irish accent, but in the US it would not be expected. Is there another product I should be looking at? I don't even know for sure what I am going to do with it yet, but I am certain I'll think of something. This is too cool not to use, but only if it is useful. So much depends on what you are trying to do. I just wanted to have a way to allow asterisk to talk in a demo, to show the concept. Unfortunately, showing a talking server with Cepestral's David is little like showing a prototype website: people don't always have the imagination (like we all do here :) to see what this would be like when actually done (or using a better voice in this case). People can be turned off very quickly. That's exactly why, whatever I end up doing with this, it needs to sound clear and be understandable. No one gives anything a second chance :( Jim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral
On 7/10/05, Jim Archer [EMAIL PROTECTED] wrote: Thanks William and John, I'll look again for that download. Comments below... --On Sunday, July 10, 2005 1:50 PM +0200 Wilson Pickett [EMAIL PROTECTED] wrote: FWIW? I bought that voice and I find it amusing, but not ready for prime time. I had it read articles from a publication and it was ludicrous. I can understand the people talking about ATT, I think I heard a demo that was very convincing. What is ATT? Is it another text to speech engine? I installed Festival a ATT Natural voices seem to be pretty good. You can hear samples here: http://www.wizzardsoftware.com/att_desktop.php . The Rich voice I think sounds the best. They are a little better than the Cepstral voices. But the Cepstral voices are vey good also. Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cepstral partnership with Digium
I just read about the partnership but was wondering what is actually going to happen? Is asterisk going to be bundled with cepstral voices for free :)? Or whats the deal? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral partnership with Digium
You will be able to purchase Cepstral voices from Digium just like you dor for G729 already. I would guess it's 1 way to show the power of asterisk by putting all the TTS orders thru a company such as Digium. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cepstral integration with * using AGI?
Hi, I've looked at the Wiki for this, have seen the Swift.agi details, but has anyone got a current script for Cepstral and an example of integraton in * please? I'm a * and linux newbie, so please be gentle ;-) Thanks John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cepstral integration with * using AGI?
On Monday January 24 2005 3:29 pm, John Middleton wrote:
Hi, I've looked at the Wiki for this, have seen the Swift.agi
details, but has anyone got a current script for Cepstral and an
example of integraton in * please?
I'm a * and linux newbie, so please be gentle ;-)
Thanks
John
I just put swift.agi in agi-bin and used a c++ script to do a db
look-up in postgres for the information that i wanted read to the
user, i.e user name and other info based on caller id number. I pass
calleridnum to the c++ script and then use SETVAR in the script to
get the info back and read it to the user. recfound is set to 1 if
any record in the db.
Extensions .conf look somewhat like this:
[answerMain]
exten = s,1,Ringing() ; Send Ring tone to caller
exten = s,2,Wait,5 ; Wait a 5 seconds to get a
ring or two
exten = s,3,Answer ; Answer the line
exten = s,4,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten = s,5,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
exten = s,6,PrivacyManager
exten = s,7,agi,c++script.cpp|${CALLERIDNUM}; //does db lookup
exten = s,8,GoToIf($[${RECFOUND} 0]?9:17); // if found a record
exten = s,9,agi,swift.agi|Welcome to the Reservation System. We will
be placing a reservation for ${varname}.;
exten = s,10,read(foo,static recording,1); //wait for user input of 1
digit
I us QT for writting the script but it is just a simple C++ script,
subject for different mail list.
John M
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Re: [Asterisk-Users] cepstral integration with * using AGI? -sent last responce to soon stupid me
On Monday January 24 2005 3:29 pm, John Middleton wrote:
Hi, I've looked at the Wiki for this, have seen the Swift.agi
details, but has anyone got a current script for Cepstral and an
example of integraton in * please?
I'm a * and linux newbie, so please be gentle ;-)
Thanks
John
I just put swift.agi in agi-bin and used a c++ script to do a db
look-up in postgres for the information that i wanted read to the
user, i.e user name and other info based on caller id number. I pass
calleridnum to the c++ script and then use SETVAR in the script to
get the info back and read it to the user. recfound is set to 1 if
any record in the db.
Extensions .conf look somewhat like this:
[answerMain]
exten = s,1,Ringing() ; Send Ring tone to caller
exten = s,2,Wait,5 ; Wait a 5 seconds to
get a ring or two
exten = s,3,Answer ; Answer the line
exten = s,4,DigitTimeout,5 ; Set Digit Timeout to 5
seconds
exten = s,5,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
exten = s,6,PrivacyManager
exten = s,7,agi,c++script.cpp|${CALLERIDNUM}; //does db lookup
exten = s,8,GoToIf($[${RECFOUND} 0]?9:17); // if found a record
exten = s,9,agi,swift.agi|Welcome to the Reservation System. We will
be placing a reservation for ${varname}.;
exten = s,10,read(foo,static recording,1); //wait for user input of 1
digit
I us QT for writting the script but it is just a simple C++ script,
subject for different mail list. but here is an example.
#include qsqldatabase.h
#include qdatatable.h
#include qsqlcursor.h
#include qsqlquery.h
#include qstring.h
#include stdio.h
#include qapplication.h
#include iostream
#include qregexp.h
#include qdatetime.h
#include qprocess.h
using namespace std;
#define DRIVER QPSQL7/* PostgreSQL Driver*/
#define DATABASE DBNAME /* the name of the database */
#define USER jmillican /* user name with
appropriate rights */
#define PASSWORD**/* password for USER */
#define HOST 127.0.0.1 /*host on which the database
is running */
QSqlDatabase * db ;
int main( int argc, char **argv)
{
bool useGUI = false;
QApplication a( argc, argv, useGUI);
setlinebuf(stdout);
setlinebuf(stderr);
QString callid;
QString astvar;
callid = a.argv()[1]; //get caller id from asterisk
QSqlDatabase * db = QSqlDatabase::addDatabase( DRIVER );
db-setDatabaseName( DATABASE );
db-setUserName( USER );
db-setPassword( PASSWORD );
db-setHostName( HOST );
if (!db-open())
{
fputs(db not open \n,stderr);
}
// get cust_id and stuff from callid
QSqlQuery query;
query.prepare(select SQL QUERY HERE);
query.bindValue(:phone,callid);
query.exec();
query.next();
QString strCustId = query.value(0).toString();
QString strsomeId = query.value(1).toString();
QString strsomeName;
QVariant vSize = query.size();
QString strSize = vSize.toString();
strsomeName = query.value(2).toString();
strsomeName = strBoatName.replace( ,,);
if (query.size() = 1) //send all info back to asterisk
{
fprintf(stdout,EXEC SETVAR CUSTID= + strCustId + \n);
fprintf(stdout,EXEC SETVAR BOATNAME= + strsomeName + \n);
fprintf(stdout,EXEC SETVAR BOATID= + strsomeId + \n);
fprintf(stdout,EXEC SETVAR RECFOUND= + strSize + \n);
}
else
{
fprintf(stdout,EXEC SETVAR RECFOUND=0 \n);// if no record
found
}
db-close();
return 0;
If any one sees this as a bad example please say so with comment on
how to make it better.
John M
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Re: [Asterisk-Users] cepstral integration with * using AGI?
Quoting John Middleton [EMAIL PROTECTED]: Hi, I've looked at the Wiki for this, have seen the Swift.agi details, but has anyone got a current script for Cepstral and an example of integraton in * please? It's been a while since I've fiddled around with it, but it should work like this: exten= s,1,Answer exten= s,2,agi(swift.agi|Hello. This is shanes boat calling.) exten= s,3,agi(swift.agi|Shane will be going out on the boat soon.) exten= s,4,agi(swift.agi|Shane will be out on the lake in uproximatly 45 minutes.) exten= s,5,agi(swift.agi|If you would like to go for a ride you should be able to meet at sun sets in Wyzeta.) exten= s,6,agi(swift.agi|To hear this message again. Touch one.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cepstral voices
Was hoping someone could point me in the right direction for getting the linux version of cepstral voices working with Asterisk. The cepstral site states that the voices work with asterisk, but I haven't been able to find anything with google or in the handbook on this. Thanks in advance, Bruce ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cepstral voices
I have successfully used Cepstral voices with asterisk. I have had better luck with integrating in by invoking Cepstral through at the command line (rather than using the Festival command from within asterisk), then just use the outputted file from your asterisk script. I have all of this automated so that it occurs as a result of actions defined in my extensions.conf file. Cepstral offers a fully functional download for you to test your architecture - and I was able to get everything running before I purchased the voice. The downloaded version has a clause at the end saying something about demo from Cepstral. Please clarify your question if you are pursuing a specific implementation path, and I'll do my best to help. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bruce Yount Sent: Tuesday, December 07, 2004 9:58 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cepstral voices Was hoping someone could point me in the right direction for getting the linux version of cepstral voices working with Asterisk. The cepstral site states that the voices work with asterisk, but I haven't been able to find anything with google or in the handbook on this. Thanks in advance, Bruce ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cepstral voices
I have successfully used Cepstral voices with asterisk. I have had better luck with integrating in by invoking Cepstral through at the command line (rather than using the Festival command from within asterisk), then just use the outputted file from your asterisk script. I have all of this automated so that it occurs as a result of actions defined in my extensions.conf file. Cepstral offers a fully functional download for you to test your architecture - and I was able to get everything running before I purchased the voice. The downloaded version has a clause at the end saying something about demo from Cepstral. Please clarify your question if you are pursuing a specific implementation path, and I'll do my best to help. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bruce Yount Sent: Tuesday, December 07, 2004 9:58 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cepstral voices Was hoping someone could point me in the right direction for getting the linux version of cepstral voices working with Asterisk. The cepstral site states that the voices work with asterisk, but I haven't been able to find anything with google or in the handbook on this. Thanks in advance, Bruce ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral voices
You should familiarize yourself with the Wiki. http://www.voip-info.org/tiki-index.php?page=Cepstral Take a gander at the See also's. On Tue, 7 Dec 2004 20:57:36 -0600, Bruce Yount [EMAIL PROTECTED] wrote: Was hoping someone could point me in the right direction for getting the linux version of cepstral voices working with Asterisk. The cepstral site states that the voices work with asterisk, but I haven't been able to find anything with google or in the handbook on this. Thanks in advance, Bruce ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Is it something someone said, was it something someone said? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cepstral available
I noticed that the linux version of Cepstral is now available. however its name is now swift, not theta. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral
On 09/09/2004 at 18:48 Josh Roberson wrote: I wrote cepstral regarding this at the beginning of the week, thought it might be relevant to post the reply: Thanks for contacting us. Our Linux package is off the site right now because we are releasing a new version, 3.02, next week. This is an incremental release. The major update of this version is a new Linux SDK. Please check back with us in 6-7 days and we should have what you're looking for. We appreciate your patience. -Craig Now hopefully, they'll hold up to it and release the new Linux SDK in a week or so... -twisted This *may* be related to my original app_cepstral that can't be integrated into CVS because of the licencing. bkw had a chat with them, iirc about making parts gpl, to solve this 'issue'.. perhaps they've done it (are doing it)... only time will tell Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cepstral
How do you get Cepstral working, they only offer windows versions. do I have to complie it to linux? http://www.cepstral.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral
On Thu, 09 Sep 2004 13:59:20 -0600, TELUX [EMAIL PROTECTED] wrote: How do you get Cepstral working, they only offer windows versions. do I have to complie it to linux? http://www.cepstral.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users They used to sell Linux versions of their voices. Now they don't seem to. Another good company bites the dust, I guess... Rob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral
On Thu, 2004-09-09 at 14:59, TELUX wrote: How do you get Cepstral working, they only offer windows versions. do I have to complie it to linux? http://www.cepstral.com They have a linux version for purchase on their web site. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral
On Thu, 09 Sep 2004 15:49:48 -0500, Eric Wieling [EMAIL PROTECTED] wrote: On Thu, 2004-09-09 at 14:59, TELUX wrote: How do you get Cepstral working, they only offer windows versions. do I have to complie it to linux? http://www.cepstral.com They have a linux version for purchase on their web site. Well, they used to, because I bought a couple voices for Linux. But they don't now. They do say 'coming soon'... Shortly after purchasing Emily, I inquired about the 'newsreader Kevin' voice that was on their demo page, but not available for sale or download. I don't remember exactly when they said it should be available, but it should have been by now. It would have been great for reading weather reports and that sort of thing... Rob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral
Quoting Jerry Geis [EMAIL PROTECTED]: Cepstral offers Linux versions. Just contact them. http://www.cepstral.com/cgi-bin/downloads?page=voices Note that you can not download any Linux versions from that page. They changed something a while back. Released a new TTS engine for Windows and Windows CE, but have not as of yet released it for Linux. I have an old version of the program called theta and I have the Frank voice which works well. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral
I wrote cepstral regarding this at the beginning of the week, thought it might be relevant to post the reply: Thanks for contacting us. Our Linux package is off the site right now because we are releasing a new version, 3.02, next week. This is an incremental release. The major update of this version is a new Linux SDK. Please check back with us in 6-7 days and we should have what you're looking for. We appreciate your patience. -Craig Now hopefully, they'll hold up to it and release the new Linux SDK in a week or so... -twisted Shane Young wrote: Quoting Jerry Geis [EMAIL PROTECTED]: Cepstral offers Linux versions. Just contact them. http://www.cepstral.com/cgi-bin/downloads?page=voices Note that you can not download any Linux versions from that page. They changed something a while back. Released a new TTS engine for Windows and Windows CE, but have not as of yet released it for Linux. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cepstral
Hi, The song with linux pakage in one week I get it 5 weeks ago, 3 weeks ago and this week :( Best regards, Chris HARIGA -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Josh Roberson Sent: Thursday, September 09, 2004 7:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cepstral I wrote cepstral regarding this at the beginning of the week, thought it might be relevant to post the reply: Thanks for contacting us. Our Linux package is off the site right now because we are releasing a new version, 3.02, next week. This is an incremental release. The major update of this version is a new Linux SDK. Please check back with us in 6-7 days and we should have what you're looking for. We appreciate your patience. -Craig Now hopefully, they'll hold up to it and release the new Linux SDK in a week or so... -twisted Shane Young wrote: Quoting Jerry Geis [EMAIL PROTECTED]: Cepstral offers Linux versions. Just contact them. http://www.cepstral.com/cgi-bin/downloads?page=voices Note that you can not download any Linux versions from that page. They changed something a while back. Released a new TTS engine for Windows and Windows CE, but have not as of yet released it for Linux. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral TTS Code
Your website is refusing connections at the moment. Or more properly I should say I get Connection refused when I try to access the Cepstral link you posted earlier today to the Asterisk-users list. FYI. Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cepstral TTS Code
Feedback for the list. I compiled Andy's code. Installation went well (except for me misspellng something in the dialplan) with no problems. The Application works great. Will run down Brian's and give it a try too. Robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral TTS Code
[EMAIL PROTECTED] wrote: Feedback for the list. I compiled Andy's code. Installation went well (except for me misspellng something in the dialplan) with no problems. The Application works great. Will run down Brian's and give it a try too. Hope you can do us a HOWTO. Cepstral would be a major win IMO compared to Festival. I use Frank, and even though he sounds a bit effete, my customers love him. I currently generate static GSMs and then play them. Being able to do it inside asterisk would be way cool. BTW what is Andy's code? Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral TTS Code
Andy's code and my code are the same code basically. I cleaned up a few things and added the noanswer option. Other than that Andy did all of the hard work. bkw On Wed, 4 Feb 2004, Brian Capouch wrote: [EMAIL PROTECTED] wrote: Feedback for the list. I compiled Andy's code. Installation went well (except for me misspellng something in the dialplan) with no problems. The Application works great. Will run down Brian's and give it a try too. Hope you can do us a HOWTO. Cepstral would be a major win IMO compared to Festival. I use Frank, and even though he sounds a bit effete, my customers love him. I currently generate static GSMs and then play them. Being able to do it inside asterisk would be way cool. BTW what is Andy's code? Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral TTS Code
Hi Brian, Andy's code and my code are the same code basically. I cleaned up a few things and added the noanswer option. Other than that Andy did all of the hard work. is cepstral a special tts-api, or does this mean, we can use every windows(tm) tts-engine on the market...? Even ATT Natural Voices...? Can we allready test this app, or is this a closed source thingy...? Greez Andreas _ Surf the net and talk on the phone with Xtra Jetstream @ http://www.xtra.co.nz/products/0,,5803,00.html ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral TTS Code
No it uses the linux theta libs and header files. bkw On Wed, 4 Feb 2004, Andreas Anderson wrote: Hi Brian, Andy's code and my code are the same code basically. I cleaned up a few things and added the noanswer option. Other than that Andy did all of the hard work. is cepstral a special tts-api, or does this mean, we can use every windows(tm) tts-engine on the market...? Even ATT Natural Voices...? Can we allready test this app, or is this a closed source thingy...? Greez Andreas _ Surf the net and talk on the phone with Xtra Jetstream @ http://www.xtra.co.nz/products/0,,5803,00.html ! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cepstral TTS Code
http://asterisk.bkw.org/other/cepstral.tar.gz bkw On Wed, 4 Feb 2004, Brian Capouch wrote: I'm prolly showing my ignorance here, but where *is* this code? I've done a search at the bugs site and it came up dry. It's not in the CVS contrib tree. Don't know where else to look. Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
