Hi all - In case anyone's interested, I've documented how we interfaced FS with Lumenvox via MRCP using FS' event socket and unicast interfaces and a bit of Perl here: http://www.softivr.com/wiki/index.php/FreeSWITCH_MRCP_in_Perl Three surprises: that it worked at all, that it works quite well and that it was really quite easy to do. One thing I'm looking for: has anyone written a module which attaches a bug to an audio stream and forwards the audio as RTP to a specified IP/port to just allow audio to be tapped off a call and sent somewhere else to be listened to? Cheers -- Dave -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk
On Mon, Jan 12, 2009 at 4:49 PM, David Knell <dave@3c.co.uk> wrote: In case anyone's interested, I've documented how we interfaced FS with
Lumenvox via MRCP using FS' event socket and unicast interfaces and a bit of Perl here: http://www.softivr.com/wiki/index.php/FreeSWITCH_MRCP_in_Perl
Three surprises: that it worked at all, that it works quite well and that it was really quite easy to do.
aka the FS motto ... "it works, its works quite well, its really quite easy ... " -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | peace be upon you ...
Great! I hope you will try doing Voxeo's Prophecy next as well ;-) Thanks Dave. Mark. -----Original Message----- From: David Knell <dave@3c.co.uk> To: freeswitch-users@lists.freeswitch.org Sent: Mon, 12 Jan 2009 3:49 am Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl Hi all - In case anyone's interested, I've documented how we interfaced FS with Lumenvox via MRCP using FS' event socket and unicast interfaces and a bit of Perl here: http://www.softivr.com/wiki/index.php/FreeSWITCH_MRCP_in_Perl Three surprises: that it worked at all, that it works quite well and that it was really quite easy to do. One thing I'm looking for: has anyone written a module which attaches a bug to an audio stream and forwards the audio as RTP to a specified IP/port to just allow audio to be tapped off a call and sent somewhere else to be listened to? Cheers -- Dave -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
"With FreeSWITCH not having any supported ASR at the time of writing (with the exception of PocketSphinx), we needed something to allow us to connect it to an MRCP server to test SoftIVR's ASR functionality. After a few false starts, we implemented a simple MRCP connector using the outbound socket interface, unicast and a bit of Perl." Was mod_openmrcp not enough :) We really need someone to fund the writing of mod_unimrcp. /b On Jan 12, 2009, at 5:49 AM, David Knell wrote:
Hi all -
In case anyone's interested, I've documented how we interfaced FS with Lumenvox via MRCP using FS' event socket and unicast interfaces and a bit of Perl here: http://www.softivr.com/wiki/index.php/FreeSWITCH_MRCP_in_Perl
Three surprises: that it worked at all, that it works quite well and that it was really quite easy to do.
One thing I'm looking for: has anyone written a module which attaches a bug to an audio stream and forwards the audio as RTP to a specified IP/port to just allow audio to be tapped off a call and sent somewhere else to be listened to?
Cheers --
Dave
-- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk
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Yup, or just get pocketsphinx "tuned" up for telephony and then no one will have to bother with ASR vendors. I believe that some speech data from a good size sample for training is needed to make it more "speaker independent" and better suited for use with phone calls. I have a list of things from the Sphinx forums that would be good to have for a telephony ready PocketSphinx. There is a "wsj" database but I don't know if that's would help?? Best. Mark. -----Original Message----- From: Brian West <brian@freeswitch.org> To: freeswitch-users@lists.freeswitch.org Sent: Mon, 12 Jan 2009 9:29 am Subject: Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl "With FreeSWITCH not having any supported ASR at the time of writing (with the exception of PocketSphinx), we needed something to allow us to connect it to an MRCP server to test SoftIVR's ASR functionality. After a few false starts, we implemented a simple MRCP connector using the outbound socket interface, unicast and a bit of Perl." Was mod_openmrcp not enough :) ?We really need someone to fund the writing of mod_unimrcp. /b On Jan 12, 2009, at 5:49 AM, David Knell wrote: Hi all - In case anyone's interested, I've documented how we interfaced FS with Lumenvox via MRCP using FS' event socket and unicast interfaces and a bit of Perl here: http://www.softivr.com/wiki/index.php/FreeSWITCH_MRCP_in_Perl Three surprises: that it worked at all, that it works quite well and that it was really quite easy to do. One thing I'm looking for: has anyone written a module which attaches a bug to an audio stream and forwards the audio as RTP to a specified IP/port to just allow audio to be tapped off a call and sent somewhere else to be listened to? Cheers -- Dave -- David Knell, Director, 3C Limited T: 020 8114 8901 ?F: 020 3002 7257 ?M: 001 415 630 3031 http://www.3c.co.uk _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org = _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Pocketsphinx works great for telephony.. just don't load 10000 word dictionary or grammar :P the pizza demo uses it.. and it works great from every phone I have tested it with... Rome wasn't built in a day and we need more people that have the skills to really build a general purpose acoustical model that works in more situations. /b On Jan 12, 2009, at 11:46 AM, mszlazak@aol.com wrote:
Yup, or just get pocketsphinx "tuned" up for telephony and then no one will have to bother with ASR vendors.
I believe that some speech data from a good size sample for training is needed to make it more "speaker independent" and better suited for use with phone calls. I have a list of things from the Sphinx forums that would be good to have for a telephony ready PocketSphinx. There is a "wsj" database but I don't know if that's would help??
Best. Mark.
That's not the opinion of Nickolay S. from the Sphinx forums. He didn't think it was telephony ready but you implied something similar in a past email. Also, I got a similar impression with the pizza demo as it came with FS. Instead I tried Voxeo's Prophecy as per your recommendation and found it worked better. As I understand it, pocketsphinx and sphinx (3 & 4) are very good but need adapting and training for there various uses. So, why bother with LumenVox, Voxeo, Nuance, etc if one could get pocketsphinx working better since it's already integrated with FS? -----Original Message----- From: Brian West <brian@freeswitch.org> To: freeswitch-users@lists.freeswitch.org Sent: Mon, 12 Jan 2009 9:55 am Subject: Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl Pocketsphinx works great for telephony.. just don't load 10000 word dictionary or grammar :P the pizza demo uses it.. and it works great from every phone I have tested it with... Rome wasn't built in a day and we need more people that have the skills to really build a general purpose acoustical model that works in more situations. /b On Jan 12, 2009, at 11:46 AM, mszlazak@aol.com wrote: Yup, or just get pocketsphinx "tuned" up for telephony and then no one will have to bother with ASR vendors. I believe that some speech data from a good size sample for training is needed to make it more "speaker independent" and better suited for use with phone calls. I have a list of things from the Sphin x forums that would be good to have for a telephony ready PocketSphinx. There is a "wsj" database but I don't know if that's would help?? Best. Mark. = _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Maybe for NON english speakers it doesn't do well but for my tests and needs it does excellent. Sphinx isn't ready thats for sure.. but PocketSphinx does great. I have PocketSphinx doing voice dial by name directory on very common and simple names. If you adapt it it can get much better. But have you called AT&T lately? I have no idea what they use but OMG it sucks... you say "NO" it doesn't understand you.. you say your account number .. it doesn't understand you... you scream curse words at it and it will take you to an agent so they can get you to the right place. Its aweful. Pocketsphinx has performed better than that on my testing. /b On Jan 12, 2009, at 5:09 PM, mszlazak@aol.com wrote:
That's not the opinion of Nickolay S. from the Sphinx forums. He didn't think it was telephony ready but you implied something similar in a past email. Also, I got a similar impression with the pizza demo as it came with FS. Instead I tried Voxeo's Prophecy as per your recommendation and found it worked better. As I understand it, pocketsphinx and sphinx (3 & 4) are very good but need adapting and training for there various uses.
So, why bother with LumenVox, Voxeo, Nuance, etc if one could get pocketsphinx working better since it's already integrated with FS?
"My god" I would LOVE it if this is really the case and would praise pocketsphinx (PS) and FS to no end. But my experience has been different. First, I tried the pizza demo with a soft phone and later by outside phone calls to my Linksys 3102 pstn-to-voip gateway. Second, I tried these two set-ups again but with Voxeo's Prophecy ASR. Both are as is and by this I mean there was no training of PocketSphinx just running the pizza demo and with Prophecy there is no training because it can't be trained. Prophecy is quite good but the FS/Pocketsphinx pizza demo isn't and I couldn't use it at a pizza join. Also, I get a much better experience when calling LumenVox and trying their pizza demo. Now, maybe Prophecy is the type of asr that doesn't require hours of training to make it speaker independent. I know that the Sphinx family are the types of ASR that do need this. So, if there is some settings for adaptation of Pocketsphinx for speaker independence then are they turned on? ? How many hours of calls to a business should an owner expect before PocketSphinx gets good enough not to scare customers away? If there are many hours needed then I could see using another ASR in the mean time, recording their calls and feeding the audio to Pocketsphinx for training, then switching to Pocketspinx once it's "tuned up." At least this way a business doesn't have to deal with a "virgin" pocketsphinx. Mark -----Original Message----- From: Brian West <brian@freeswitch.org> To: freeswitch-users@lists.freeswitch.org Sent: Mon, 12 Jan 2009 3:21 pm Subject: Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl Maybe for NON english speakers it doesn't do well but for my tests and needs it does excellent. Sphinx isn't ready thats for sure.. but PocketSphinx does great. I have PocketSphinx doing voice dial by name directory on very common and simple names. If you adapt it it can get much better. But have you called AT&T lately? I have no idea what they use but OMG it sucks... you say "NO" it doesn't understand you.. you say your account number .. it doesn't understand you... you scream curse words at it and it will take you to an agent so they can get you to the right place. Its aweful. Pocketsphinx has performed better than that on my testing. /b On Jan 12, 2009, at 5:09 PM, mszlazak@aol.com wrote:
That's not the opinion of Nickolay S. from the Sphinx forums. He didn't think it was telephony ready but you implied something similar in a past email. Also, I got a similar impression with the pizza demo as it came with FS. Instead I tried Voxeo's Prophecy as per your recommendation and found it worked better. As I understand it, pocketsphinx and sphinx (3 & 4) are very good but need adapting and training for there various uses.
So, why bother with LumenVox, Voxeo, Nuance, etc if one could get pocketsphinx working better since it's already integrated with FS?
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I'm missing something, What is your point exactly? I *just* explained that we want to support unimrcp so you can use Prophecy if you wish so we get it, there is no need to continue to complain. I tried to tell you that we are short on time and we are trying our best. We had openmrcp and the devloper discontinued the project. Now we need to get rid of it and switch to unimrcp. If you recall, you called us for consulting, then spent an hour on the phone gathering free information then proceeded to get all kinds of free help on this list using the free software we have made available to you. It's great that Prophecy is the only place you want to spend any money and I encourage you to do so we can connect you with Voxeo any time. But what else exactly do you want from us? You may want to factor in that your limited experience and particular requirements contribute to your trouble setting everything up so clearly the pocketsphinx route is not for you. (You are only the 2nd person to try it on windows for instance) I keep reading all of your emails and I am trying to understand exactly what you want from us. On Tue, Jan 13, 2009 at 12:55 AM, <mszlazak@aol.com> wrote:
"My god" I would LOVE it if this is really the case and would praise pocketsphinx (PS) and FS to no end. But my experience has been different.
First, I tried the pizza demo with a soft phone and later by outside phone calls to my Linksys 3102 pstn-to-voip gateway. Second, I tried these two set-ups again but with Voxeo's Prophecy ASR.
Both are as is and by this I mean there was no training of PocketSphinx just running the pizza demo and with Prophecy there is no training because it can't be trained.
Prophecy is quite good but the FS/Pocketsphinx pizza demo isn't and I couldn't use it at a pizza join. Also, I get a much better experience when calling LumenVox and trying their pizza demo.
Now, maybe Prophecy is the type of asr that doesn't require hours of training to make it speaker independent. I know that the Sphinx family are the types of ASR that do need this.
So, if there is some settings for adaptation of Pocketsphinx for speaker independence then are they turned on?
How many hours of calls to a business should an owner expect before PocketSphinx gets good enough not to scare customers away?
If there are many hours needed then I could see using another ASR in the mean time, recording their calls and feeding the audio to Pocketsphinx for training, then switching to Pocketspinx once it's "tuned up." At least this way a business doesn't have to deal with a "virgin" pocketsphinx.
Mark
-----Original Message----- From: Brian West <brian@freeswitch.org> To: freeswitch-users@lists.freeswitch.org Sent: Mon, 12 Jan 2009 3:21 pm Subject: Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl
Maybe for NON english speakers it doesn't do well but for my tests and
needs it does excellent. Sphinx isn't ready thats for sure.. but
PocketSphinx does great.
I have PocketSphinx doing voice dial by name directory on very common
and simple names. If you adapt it it can get much better. But have
you called AT&T lately? I have no idea what they use but OMG it
sucks... you say "NO" it doesn't understand you.. you say your account
number .. it doesn't understand you... you scream curse words at it
and it will take you to an agent so they can get you to the right
place. Its aweful. Pocketsphinx has performed better than that on my
testing.
/b
On Jan 12, 2009, at 5:09 PM, mszlazak@aol.com wrote:
That's not the opinion of Nickolay S. from the Sphinx forums. He
didn't think it was telephony ready but you implied something
similar in a past email. Also, I got a similar impression with the
pizza demo as it came with FS. Instead I tried Voxeo's Prophecy as
per your recommendation and found it worked better. As I understand
it, pocketsphinx and sphinx (3 & 4) are very good but need adapting
and training for there various uses.
So, why bother with LumenVox, Voxeo, Nuance, etc if one could get
pocketsphinx working better since it's already integrated with FS?
_______________________________________________
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Apparently you did. I was responding to comments/claims of another poster by relating my experiences and wishing that PS in FS would preform better. -----Original Message----- From: Anthony Minessale <anthony.minessale@gmail.com> To: freeswitch-users@lists.freeswitch.org Sent: Tue, 13 Jan 2009 8:58 am Subject: Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl I'm missing something, What is your point exactly? I *just* explained that we want to support unimrcp so you can use Prophecy if you wish so we get it, there is no need to continue to complain.? I tried to tell you that we are short on time and we are trying our best. We had openmrcp and the devloper discontinued the project.? Now we need to get rid of it and switch to unimrcp. If you recall, you called us for consulting, then spent an hour on the phone gathering free information then proceeded to get all kinds of free help on this list using the free software we have made available to you.? It's great that Prophecy is the only place you want to spend any money and I encourage you to do so we can connect you with Voxeo any time.? But what else exactly do you want from us? You may want to factor in that your limited experience and particular requirements contribute to your trouble setting everything up so clearly the pocketsphinx route is not for you.? (You are only the 2nd person to try it on windows for instance) I keep reading all of your emails and I am trying to understand exactly what you want from us. On Tue, Jan 13, 2009 at 12:55 AM, <mszlazak@aol.com> wrote: "My god" I would LOVE it if this is really the case and would praise pocketsphinx (PS) and FS to no end. But my experience has been different. First, I tried the pizza demo with a soft phone and later by outside phone calls to my Linksys 3102 pstn-to-voip gateway. Second, I tried these two set-ups again but with Voxeo's Prophecy ASR. Both are as is and by this I mean there was no training of PocketSphinx just running the pizza demo and with Prophecy there is no training because it can't be trained. Prophecy is quite good but the FS/Pocketsphinx pizza demo isn't and I couldn't use it at a pizza join. Also, I get a much better experience when calling LumenVox and trying their pizza demo. Now, maybe Prophecy is the type of asr that doesn't require hours of training to make it speaker independent. I know that the Sphinx family are the types of ASR that do need this. So, if there is some settings for adaptation of Pocketsphinx for speaker independence then are they turned on? ? How many hours of calls to a business should an owner expect before PocketSphinx gets good enough not to scare customers away? If there are many hours needed then I could see using another ASR in the mean time, recording their calls and feeding the audio to Pocketsphinx for training, then switching to Pocketspinx once it's "tuned up." At least this way a business doesn't have to deal with a "virgin" pocketsphinx. Mark -----Original Message----- From: Brian West <brian@freeswitch.org> To: freeswitch-users@lists.freeswitch.org Sent: Mon, 12 Jan 2009 3:21 pm Subject: Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl Maybe for NON english speakers it doesn't do well but for my tests and needs it does excellent. Sphinx isn't ready thats for sure.. but PocketSphinx does great. I have PocketSphinx doing voice dial by name directory on very common and simple names. If you adapt it it can get much better. But have you called AT&T lately? I have no idea what they use but OMG it sucks... you say "NO" it doesn't understand you.. you say your account number .. it doesn't understand you... you scream curse words at it and it will take you to an agent so they can get you to the right place. Its aweful. Pocketsphinx has performed better than that on my testing. /b On Jan 12, 2009, at 5:09 PM, mszlazak@aol.com wrote:
That's not the opinion of Nickolay S. from the Sphinx forums. He
didn't think it was telephony ready but you implied something
similar in a past email. Also, I got a similar impression with the
pizza demo as it came with FS. Instead I tried Voxeo's Prophecy as
per your recommendation and found it worked better. As I understand
it, pocketsphinx and sphinx (3 & 4) are very good but need adapting
and training for there various uses.
So, why bother with LumenVox, Voxeo, Nuance, etc if one could get
pocketsphinx working better since it's already integrated with FS?
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org A Good Credit Score is 700 or Above. See yours in just 2 easy steps! _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
You should be careful who you ask and who's opinion you weigh against. Just because you go ask some other guy something does not make it gospel. These are the basic facts: ASR/TTS is one of the many areas where we have done a huge amount of pro bono work to get people interested but there is only so much we can do as demo's and examples. We chose to add pocketsphinx because it's free and gives people a way to test. We don't have any monitary support from any of the commerical ASR/TTS providers nor do we have any support from any users who intend to improve the ASR/TTS support. Therefore given the endless hours we spend working on FS we have to decide what to work on and what to put aside. We are currently looking at trying get unimrcp working but again, nobody is willing to pay so the timeline is extended to our spare time. We want to support every ASR/TTS interface we can because it helps to expand our flexablilty but it simply will take a while. On Mon, Jan 12, 2009 at 5:09 PM, <mszlazak@aol.com> wrote:
That's not the opinion of Nickolay S. from the Sphinx forums. He didn't think it was telephony ready but you implied something similar in a past email. Also, I got a similar impression with the pizza demo as it came with FS. Instead I tried Voxeo's Prophecy as per your recommendation and found it worked better. As I understand it, pocketsphinx and sphinx (3 & 4) are very good but need adapting and training for there various uses. So, why bother with LumenVox, Voxeo, Nuance, etc if one could get pocketsphinx working better since it's already integrated with FS?
-----Original Message----- From: Brian West <brian@freeswitch.org> To: freeswitch-users@lists.freeswitch.org Sent: Mon, 12 Jan 2009 9:55 am Subject: Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl
Pocketsphinx works great for telephony.. just don't load 10000 word dictionary or grammar :P the pizza demo uses it.. and it works great from every phone I have tested it with... Rome wasn't built in a day and we need more people that have the skills to really build a general purpose acoustical model that works in more situations. /b On Jan 12, 2009, at 11:46 AM, mszlazak@aol.com wrote:
Yup, or just get pocketsphinx "tuned" up for telephony and then no one will have to bother with ASR vendors.
I believe that some speech data from a good size sample for training is needed to make it more "speaker independent" and better suited for use with phone calls. I have a list of things from the Sphinx forums that would be good to have for a telephony ready PocketSphinx. There is a "wsj" database but I don't know if that's would help??
Best. Mark.
=
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Hi Brian,
"With FreeSWITCH not having any supported ASR at the time of writing (with the exception of PocketSphinx), we needed something to allow us to connect it to an MRCP server to test SoftIVR's ASR functionality. After a few false starts, we implemented a simple MRCP connector using the outbound socket interface, unicast and a bit of Perl."
Was mod_openmrcp not enough :) We really need someone to fund the writing of mod_unimrcp. mod_openmrcp is (from our testing) badly broken: it segfaults on the second call; it is (in my opinion) unnecessarily baroque and it is no longer supported.
Cheers -- Dave
Putting some effort behind unimrcp would be the best case at this point right? /b On Jan 12, 2009, at 12:20 PM, David Knell wrote:
mod_openmrcp is (from our testing) badly broken: it segfaults on the second call; it is (in my opinion) unnecessarily baroque and it is no longer supported.
Cheers --
Dave
This is great. On Jan 12, 2009, at 6:49 AM, David Knell wrote:
Hi all -
In case anyone's interested, I've documented how we interfaced FS with Lumenvox via MRCP using FS' event socket and unicast interfaces and a bit of Perl here: http://www.softivr.com/wiki/index.php/FreeSWITCH_MRCP_in_Perl
Three surprises: that it worked at all, that it works quite well and that it was really quite easy to do.
One thing I'm looking for: has anyone written a module which attaches a bug to an audio stream and forwards the audio as RTP to a specified IP/port to just allow audio to be tapped off a call and sent somewhere else to be listened to?
Cheers --
Dave
-- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk
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participants (6)
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Andrew Gilbert
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Anthony Minessale
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Brian West
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David Knell
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mszlazak＠aol.com
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Wasim Baig