Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl
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 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?
Mark, Sorry I haven't had much time to help with this either. But Anthony is offering good advice here. You are either going to have to work out what is going on at SIP/SDP/RTP level through logs and wireshark, or opt for a separate ip space. Another option (besides virtual ips) is VMWare or VirtualBox, although VMWare is probably easier to setup and bridge naturally to your host. Vm's are just so easy anymore and it definitely seems like you are going against the grain right now. Also - realizing you got here because of the need for ASR. I do have the Lumenvox license, and I was able to compile the module out of SVN. I have not tested anything yet. If things go well I should have some time after the 25th for this. My goal would be to get pizza or something akin to work. Andy On Dec 22, 2008, at 10:24 AM, Anthony Minessale wrote: I don't really know what your problem is. I just saw you ask 3 times for help and tried to offer a suggestion. if you start FS with TPORT_LOG=1 you can see all the sip messages in the console and you could also run wireshark to look at a packet capture. If you use the same IP for media on the same box for 3 programs at once you may end up with 2 applictions choosing the same media port etc. It's just a good practice to run every voip program on it's own IP. On Mon, Dec 22, 2008 at 12:44 AM, mszla...@aol.com wrote: Hi Anthony, I actually suggested adding IP's to a Voxeo-Prophecy support person before but they thought that could be problematic. I went along with the earlier warning but now you have suggested it again. What makes everything on the same box tricky? Also, the thing that surprises me a bit is that bypass-media works but proxy-media or the default doesn't. Would you be kind enough to elaborate. Thanks. Mark. -Original Message- From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Sun, 21 Dec 2008 2:49 pm Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work? Try adding more ip to your box and give each thing it's own dedicated virtual IP. Doing everything on the same box can be tricky. On Sat, Dec 20, 2008 at 2:17 AM, mszla...@aol.com wrote: With the firewall ON or OFF the problem still remains. I've tried 3 different set-ups in a dial plan extension. 1. With only action application=set data=proxy_media=true/ before bridging. 2. With only action application=set data=bypass_media=true/ before bridging. 3. Neither of the above in the extension. Only 2 with bypass-media=true gets the audio across endpoints. Help :-) -Original Message- From: mszla...@aol.com To: freeswitch-users@lists.freeswitch.org Sent: Fri, 19 Dec 2008 11:30 am Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work? With the firewall ON or OFF the problem still remains. I've tried 3 different set-ups in a dial plan extension. 1. With only action application=set data=proxy_media=true/ before bridging. 2. With only action application=set data=bypass_media=true/ before bridging. 3. Neither of the above in the extension. Only 2 with proxy-media=true gets the audio across endpoints. Help :-) 0A -Original Message- From: Michael Jerris m...@jerris.com To: freeswitch-users@lists.freeswitch.org Sent: Fri, 19 Dec 2008 7:49 am Subject: Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work? It gives me the impression there is something wrong with your firewall running on the box. Mike On Dec 19, 2008, at 3:03 AM, mszla...@aol.com wrote: I find it strange that I can have to endpoints get audio went using bypass media mode but the audio fails to go across endpoints if I use proxy media mode. I'm trying to pass audio internally on the same machine between endpoints and have be advis ed that a reason the audio may fail to be passed is because there is some RTP timing and IP address/port issues. However, FS has no problem connecting ports if i change the mode to bypass media. This gives me the impression that something is wrong with FS proxy media mode. Any comments? Listen to 350+ music, sports, news radio stations – including songs for the holidays – FREE while you browse. Start Listening Now! ___ 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 Listen to 350+ music, sports, news radio stations –
Re: [Freeswitch-users] how to force a MINIMUM call duration
Don't want the tone to be wrong here, but this makes no sense. Carriers surcharge like this precisely to guard against call center, predictive and other mass outbound calling scenarios. It just doesn't make since, math wise, that individuals hanging up on voice mail are going to significantly impact overall ACD stats, etc. So unless you have a very strange set of use cases or are pushing another category of traffic (ie call center) that skews you overall relationship with the carrier - I would go back and re-negotiate your arrangement. Yes, FS is a b2bua and all is possible. But it is probably a better use of time to approach this as a business issue. My 2 cents. On Dec 9, 2008, at 8:38 AM, Frank @ Impact wrote: How can FS force a Minimum call duration for a FS caller (someone calling out of FS)? We have a carrier that penalizes us with a surcharge for short duration calls (sound familiar?). So when a FS caller (not a call center or predictive dialer) calls a cell phone and gets a ring tone or calls an answering machine, the FS caller hangs up because they do not want to leave a message. But they do this in less then a few seconds after the call is answered. This becomes a short duration call and bang the surcharge applies. It is actually cheaper to pay for a longer call time (6 seconds in this case) and avoid the short duration surcharge. But the FS caller does not know this. So, how can FS hold the connection to the called party open for at least the minimum amount of time I need to avoid the short call charge… even though my FS caller has already hung up the phone on his end? I would like to do this in the xml dialplanif possible. Thanks -Frank ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Question Regarding ASR/TTS and CMU OSS Projects
Curious if anyone has practical real world input on training CMU based ASR engines (Sphinx, PocketSphinx) and / or creating and tuning voices for the TTS related components. Just trying to understand how hard it is, what the realistic gap is to use these tools in real world applications. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] how to force a MINIMUM call duration
What do your records say? Ie do they balance to what the carrier claims? You should at a minimum have macro level data to confirm against. 27% seems high, but even at that level if you assume your remaining population is normal you are still no where close to call center / predictive traffic in the overall sense. For example, 2 minutes ACD on the normal population is still almost 90 seconds overall. Compare this to outbound call centers that might have an overall ACD in the 10-30 second range and have well over 50%, probably much higher, short duration. I would tell your carrier to stop being silly, or find another one. I am unsure you can do it just in the dialplan, but it is a somewhat trivial app. The issue is it is difficult to safely avoid scenarios where leg B might actually be a real person, talking to dead air. This is not good citizenship. It breaks implicit assumptions about network behavior and is unfair to end users. It is illegal if applied to a predictive scenario. On Dec 9, 2008, at 5:35 PM, Frank @ Impact wrote: On our last bill, the carrier said we had 27% short duration calls (maybe they are wrong but it was on the bill). It is definitely not call center. But these callers hangup as soon as they hear answer machine or most of the time a ring back tone from cell phone. This class of caller will call a cell phone, hear the ring back, hangup right away and then call back another 2 minutes later and repeat the cycle. So, if I have to make it work the way I suggested (hold the connection open for at least the minimum time, how might you suggest I do it in the dial plan? -Original Message- Don't want the tone to be wrong here, but this makes no sense. Carriers surcharge like this precisely to guard against call center, predictive and other mass outbound calling scenarios. It just doesn't make since, math wise, that individuals hanging up on voice mail are going to significantly impact overall ACD stats, etc. So unless you have a very strange set of use cases or are pushing another category of traffic (ie call center) that skews you overall relationship with the carrier - I would go back and re- negotiate your arrangement. Yes, FS is a b2bua and all is possible. But it is probably a better use of time to approach this as a business issue. My 2 cents. On Dec 9, 2008, at 8:38 AM, Frank @ Impact wrote: How can FS force a Minimum call duration for a FS caller (someone calling out of FS)? We have a carrier that penalizes us with a surcharge for short duration calls (sound familiar?). So when a FS caller (not a call center or predictive dialer) calls a cell phone and gets a ring tone or calls an answering machine, the FS caller hangs up because they do not want to leave a message. But they do this in less then a few seconds after the call is answered. This becomes a short duration call and bang the surcharge applies. It is actually cheaper to pay for a longer call time (6 seconds in this case) and avoid the short duration surcharge. But the FS caller does not know this. So, how can FS hold the connection to the called party open for at least the minimum amount of time I need to avoid the short call charge… even though my FS caller has already hung up the phone on his end? I would like to do this in the xml dialplanif possible. Thanks -Frank ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to force FS using the hosts entry in outgoing gateway calls
Jon, You should also be able to do a 'order hosts,bind' in /etc/hosts, no On Dec 5, 2008, at 11:43 AM, Jon Bruel wrote: For the configuration of a gateway I need to use a specific proxy domain name before the server (Covergence SBC with a BroadWorks Application Server behind) accepts calls. The twist is that the right proxy name points the wrong IP-address (the voicemail server for the account). I have tried to overrule this by adding a host entry (Linux). When I ping to the domain name I get the right address (the one from the host table), but the FS uses the address from the DNS lookup, not the address from the host table. What can I do to force the FS using the entry from the host table? Thanks /Jon. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Problems with Mod_openMRCP
Mark and David, I am willing to help some with testing here as well, if you need it. Ping me directly or we can get on the IRC. I am on Mac OS, but have readily available vm's with Debian, etc. I also have Prophecy. I have a general interest in an ASR solution as well. Voxeo is great, but using it as an MRCP proxy seems odd. As a full fledged VXML solution it is great, if you can afford it. But having a good ASR solution is good first step to trying to get something like OpenVXI working as well. That said, seems like a bounty or money to help FS is a better spend anyway. It is a one time cost, not a variable cost. And it goes straight to the guys doing the real work. I built unimrcp last night, it was quite straight forward. In theory, if I weren't old and my C/autoconf skills rather atrophied, it wouldn't seem like it would be that huge a deal to port/fix openmrcp to unimrcp. Finally, Anthony I was looking at the Lumenvox path as well, but got deterred by the licensing hassle. This seems to be a universal ASR issue. I would reason I can find the old module in SVN? Were they going to grant community dev licenses? Again - I am willing to volunteer to do some testing/doc at least. Andy On Dec 2, 2008, at 10:51 AM, Anthony Minessale wrote: If you can get it to break on linux I will ssh in and fix it for you. If you cannot, i can try to fix it for you over rdp but that won't be very fun. We can think about reinstating mod_lumenvox as well as another windows based asr alternative. I deleted it for the same reason we will probably delete mod_openmrcp because nobody was using it and there was no way to support it because our dev licenses had expired. Lumenvox has offered us some new dev licenses to bring it back but I would need someone to actually want it to work to put in charge of it. We will be clear about what is supported and what is not in the 1.0.2 release scheduled to be released in the near future. On Tue, Dec 2, 2008 at 12:11 AM, David Knell [EMAIL PROTECTED] wrote: Hi Anthony, mod_openmrcp was a contribution to the community by a 3rd party individual. As i have clearly stated in 2 previous emails, the man has decided to discontinue the openmrcp project. So now we are left with the remains of the module and discontinued code. This was not our decision it was his. I absolutely understand this but it's important, from a user point of view, to be able to know which bits of FS are current/supported and which aren't. Some people use it without issue which may mean that the crash you reported is windows specific and I do not have a working lab of any mrcp capbable system to try it against in unix for that matter. I have a list of work to do from here to the moon and back so on an issue like this, unless someone can hand me login credentials to some box and give me a phone number to dial to reporduce the issue, it will be a long time until we can deal with it. It's useful to know that there are people using mod_openmrcp without issue: I did ask here if anyone was a while back, and no-one fessed up. I'll give it a go on a Linux box and report back. And if you'd like a dev/test environment set up, then just tell me which one. And the question arises, should we bother working on it anymore if the lib has been abandoned and we cannot even get any support from it's author which is where the problem most likely lies. I try not to get too annoyed by these remarks about what we *ought to do* because I know people lose sight of how much of the work to support the project is done by a small group of 3 people and not the 2000 people it appears to be from the outside looking in. (I've been answering email for 4 hours now) Those guys who claim to have all that money in an offshore bank account are lying - you don't have to reply to them in future ;-) Seriously, though, I don't think it's too outrageous an idea to document what's supported and were you (for example) to have suggested that I get in touch with the contributors to the various modules, ask them what their view of its status is, condense the answers in to a list and report back, it's something I'd quite happily do. My suggestion is to pool some cash and pay the guy to make mod_unimrcp for FS that we can maintain in tree knowing the development can be supported by the original author. Quite happy to participate in that, too.. the problem is that I've a demo to do like yesterday and the timescale for mod_unimrcp is a bit on the long side for that. I'd rather not have to do it with Asterisk and Lumenvox..! Cheers -- Dave On Mon, Dec 1, 2008 at 12:51 PM, David Knell [EMAIL PROTECTED] wrote: Hi Mike, My experience is that it's somewhat broken - it took two trivial tweaks to get it to work with IBM's ASR and TTS, but there's a more intractable problem to do with memory getting overwritten (I assume that
Re: [Freeswitch-users] VxML Parser?
It looks like there are 2 viable vxml parser projects, listed here. The commetrex project is a close relative if not simply OpenVxi. The rev at CMU is older, but the one on SourceForge appears semi-current (3.4). This is also what is being used by VoiceGlue, an integration with Asterisk. The jvoicexml project is java based and doesn't look to be as far along in terms of tag support. So I spent a few minutes with OpenVXI. OpenVxi has a set of interfaces one needs to support to make a full fledged voice browser. Some are provided in the reference implementation, such as logging, etc. Others need to be provided, for example call control (VXItel.h), prompting (VXIprompt.h), etc. Seems like this would all be possible using something like event socket and glue code to implement the required OpenVXI interfaces. Call control and prompting sure seem possible that way. But I would really want someone from FS to comment on whether that makes sense. If I get more bandwidth over next few days I will try to get OpenVXI built and explore more. But again, having some input from FS devs would be great. Andy On Nov 3, 2008, at 11:57 PM, Douglas Garstang wrote: Thanks. Maybe I need to go and get my hands more dirty, but how would freeswitch interface to this? Doug. From: EdPimentl [EMAIL PROTECTED] To: freeswitch-users@lists.freeswitch.org Sent: Sunday, November 2, 2008 1:39:22 PM Subject: Re: [Freeswitch-users] VxML Parser? Here.. The most active VoiceXML projects... http://jvoicexml.sourceforge.net/ http://jvoicexml.sourceforge.net/documentation.htm http://www.commetrex.com/products/ctmiddleware/bladewarevxml.html -E ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] VxML Parser?
I am just an observer of FS. What I am impressed with is the extensibility. And I am curious about xml models (vxml, proprietary xml) vs scripting (lua, python, etc). That is why I asked. That said, sounds like the pieces are all waiting for the right motivation and resource set. I can't rule out helping, but can't promise much yet either. Lastly, curious about the not very good part or your response. Is this a reference to OpenVXI or one of it's derivatives? (and wikipbx does look cool) On Nov 2, 2008, at 11:18 PM, Michael S Collins wrote: Doug, Welcome to FreeSWITCH! BTW, I owe you a thank you because I first heard of FS from one of your posts to the asterisk users list. :) FS is definitely designed to be extensible. In fact, one night when Tony was ticked off about a post earlier in the day, he whipped up mod_yaml in three hours! However, I don't believe anyone has yet sung the virtues of VXML to the point of imploring the devs to add it, nor has anyone put up a bounty. My guess is that VXML would be added at some point. It will happen more quickly if Tony doesn't have to do it gratis if know what I mean. Hey are you still a Python guy? Just curious if you have played with mod_python or wikipbx yet. -MC Sent from my iPhone On Nov 2, 2008, at 1:15 PM, Douglas Garstang [EMAIL PROTECTED] wrote: Well, the same could be said about Asterisk and yet there are third part vxml parsers (although not very good) available for it. I have developers that want to write voice applications directly in vxml. They want to use vxml, not js, lua, python or anything else. That's a hard requirement. I'm surprised freeswitch doesn't have this available. Being able to rapidly deploy voice applications with vxml doesn't seem like an unusual requirement. Doug. From: Andrew Gilbert [EMAIL PROTECTED] To: freeswitch-users@lists.freeswitch.org Sent: Sunday, November 2, 2008 12:51:31 PM Subject: Re: [Freeswitch-users] VxML Parser? Quick answer, it's a switch/b2bua and not a vxml parser. Longer answer involves questions about what you are trying to do. If you need quick voice apps, check out the many options including js, lua, python, liverpie, etc. It is a rich set of options and much of this may be more appropriate / productive than full vxml dependent on requirements. If you have a hard requirement / need for vxml might check out voiceglue.org. This is an asterisk related project, but in theory and with some effort who knows. On Nov 2, 2008, at 12:36 PM, Douglas Garstang wrote: I'm new to freeswitch. Is there a vxml parser for freeswitch? Doug. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] VxML Parser?
Quick answer, it's a switch/b2bua and not a vxml parser. Longer answer involves questions about what you are trying to do. If you need quick voice apps, check out the many options including js, lua, python, liverpie, etc. It is a rich set of options and much of this may be more appropriate / productive than full vxml dependent on requirements. If you have a hard requirement / need for vxml might check out voiceglue.org. This is an asterisk related project, but in theory and with some effort who knows. On Nov 2, 2008, at 12:36 PM, Douglas Garstang wrote: I'm new to freeswitch. Is there a vxml parser for freeswitch? Doug. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] IVR References
No expert on this, but have you tried to update /usr/local/freeswitch/ conf/lang/en/en.xml to insure you include my-ivr.xml? So in addition to this: X-PRE-PROCESS cmd=include data=demo/*.xml/ have this as well: X-PRE-PROCESS cmd=include data=my-ivr/*.xml/ On Sep 26, 2008, at 9:03 PM, jflowers wrote: I'm having difficulty in trying to configure my own IVR (not modifying the demo_ivr files). I think I have most of it figured out but I don't understand the structure and my friends grep and find aren't helping much. I can't seem to figure out why the macros in /usr/local/freeswitch/conf/lang/en/demo/demo-ivr.xml are parsed but those in /usr/local/freeswitch/conf/lang/en/mydir/my-ivr.xml are not. I must be missing something. Any direction as to where this is specified will be greatly appreciated. Thanks. -- View this message in context: http://www.nabble.com/IVR-References-tp19698602p19698602.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
