Re: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN Updates - Bootstrap Required
I get a lot of these errors on vc++ express 2008: 59Linking... 59LINK : fatal error LNK1181: cannot open input file '..\..\..\..\w32\library\debug\freeswitchcore.lib' 59Build log was saved at file://c:\Documents and Settings\Mark Szlazak\My Documents\FreeSWITCH\src\mod\applications\mod_fsv\Win32\Debug\BuildLog.htm 59mod_fsv - 1 error(s), 1 warning(s) -Original Message- From: Michael Collins m...@freeswitch.org To: freeswitch-users@lists.freeswitch.org Sent: Fri, 22 May 2009 8:36 am Subject: Re: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN Updates - Bootstrap Required On Fri, May 22, 2009 at 7:58 AM, mszla...@aol.com wrote: What problems will a Windows user have when updating with Tortoise SVN? I haven't had a chance to test it out but what I would do is update and then rebuild solution and see how it goes. Let us know if you run into any issues. -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN Updates - Bootstrap Required
Yes I did. -Original Message- From: Michael S Collins m...@freeswitch.org To: freeswitch-users@lists.freeswitch.org freeswitch-users@lists.freeswitch.org Sent: Sat, 23 May 2009 9:42 am Subject: Re: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN Updates - Bootstrap Required Did you try clean solution as brian suggested? -MC Sent from my iPhone On May 23, 2009, at 8:33 AM, mszla...@aol.com wrote: I get a lot of these errors on vc++ express 2008: 59Linking... 59LINK : fatal error LNK1181: cannot open input file '..\..\..\..\w32\library\debug\freeswitchcore.lib' 59Build log was saved at file://c:\Documents and Settings\Mark Szlazak\My Documents\FreeSWITCH\src\mod\applications\mod_fsv\Win32\Debug\BuildLog.htm 59mod_fsv - 1 error(s), 1 warning(s) -Original Message- From: Michael Collins m...@freeswitch.org To: freeswitch-users@lists.freeswitch.org Sent: Fri, 22 May 2009 8:36 am Subject: Re: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN Updates - Bootstrap Required On Fri, May 22, 2009 at 7:58 AM, mszla...@aol.com wrote: What problems will a Windows user have when updating with Tortoise SVN? I haven't had a chance to test it out but what I would do is update and then rebuild solution and see how it goes. Let us know if you run into any issues. -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org We found the real 'Hotel California' and the 'Seinfeld' diner. What will you find? Explore WhereItsAt.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 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN Updates - Bootstrap Required
What problems will a Windows user have when updating with Tortoise SVN? -Original Message- From: Michael Collins m...@freeswitch.org To: freeswitch-users@lists.freeswitch.org freeswitch-users@lists.freeswitch.org; freeswitch-...@lists.freeswitch.org freeswitch-...@lists.freeswitch.org Sent: Thu, 21 May 2009 8:54 pm Subject: [Freeswitch-users] IMPORTANT: Latest FreeSWITCH SVN Updates - Bootstrap Required FYI, We just want to let everyone know that we have made a few updates that will require a rebootstrap. One of the key updates was a security fix for libsndfile. In this particular case it won't be possible simply to make current like you normally do. Here is a common set of commands for a typical Linux rebootstrap: cd /usr/src/freeswitch.trunk make clean svn up ./bootstrap.sh ./configure make install NOTE: if you've got the libzrtp file and you've already run the buildzrtp.sh script then be sure to use ./configure --enable-zrtp in the above operation. Thank you for your continued support of the FreeSWITCH project! -Michael S Collins http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] text to speech IVRs and MOH
Neospeech does have the best voices. I looked at Neospeech months ago and talked to a rep. Then he quoted me a price per port of over $1000.00. Looks like they really have done some big price adjustments. -Original Message- From: Saeed Ahmad saeedahmad1...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Fri, 22 May 2009 2:01 am Subject: Re: [Freeswitch-users] text to speech IVRs and MOH Thanks guys for a detailed reply specially pete. On Tue, May 19, 2009 at 7:31 PM, Peter P GMX prometheus...@gmx.net wrote: Thanks for this overwiev. One question: How does this compare to Cepstral TTS? Best regards Peter p...@privateconnect.com schrieb: I've spent the last 2-3 months on researching TTS and ASR for FS for a project. ?Best TTS depends on what you consider important. ?Also, how do you plan on using it. Here's some of the TTS engines I've run across with some pros/cons: Festivate Lite (flite) Pros: - Free (comes with FS) - simple to use - 16K voice sounds decent - Completely customizable Cons: - 8K voice sounds horrible over cell phone NeoSpeech (VoiceWare) (around $300/port for 1 voice + $75 each addl voice) Pros: - My selection for best soundig voices - Recently select by Stephen Hawkings for his voice (geek points!) - Lots of Languages supported - Free trial available Cons: - Custom C-Based API (FS interface coming soon) - Large file size (Engine + SDK + 1 Voice = 900MB) - Support is lacking (Company beed in Korean, time zone issues, etc) Nuance ($500/port for 1 voice) Pros: - Wide Variety of Products - Support MRCP - Supports ASR as well (add'l fees) - Excellent support - Free trial - Decent sounding voices at 8K and 16K - Wide range of tuning parameters Cons: - Pricey - Limited voice selection - Limited support for 64-bit linux ATT (NaturalVoice) (no pricing info available) Pros: - Big company (solid in marketplace) - Good suppport (user and developer) - ASP model means no software to maintain Cons: - ASP model incurs delay - Voices sound too digitized - Limited support for 64-bit linux Loquendo ($500/port for 1 voice + 15% addl voice) Pros: - Good sounding voices (almost as good as NeoSpeech) - Wide variety of languages - Excellent support - Has free 30 day trial - Supports MRCP - Support ASR and Voice Recognition as well. (add'l fees) - Small footprint ( 150MB) Cons: - Pricey - Complicated install process - Limited management/tuning capabilities In the end, it was down to NeoSpeech or Loquendo for our application. ?We are currently running tests with NeoSpeech and assuming all goes well, we will select them. ?Though don't let that color your opinion too much after several focus groups we discovered the most important element in the equation is does your customer/boss like the sound of the voices, and that is a completely subjective decision. -pete ? ? Original Message ? ? Subject: [Freeswitch-users] text to speech IVRs and MOH ? ? From: Saeed Ahmad saeedahmad1...@gmail.com ? ? Date: Tue, May 19, 2009 12:40 am ? ? To: freeswitch-users@lists.freeswitch.org ? ? Hi all, ? ? Could you guys recommend me any online text to speech IVR software ? ? which works OK with FS. i am using ATT site and for some IVRs i ? ? get sample rate errors. Also some resource to download more MOH ? ? wav files. ? ? Many thanks ? ? ? ? ___ ? ? Freeswitch-users mailing list ? ? Freeswitch-users@lists.freeswitch.org ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users ? ? UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users ? ? http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Re: [Freeswitch-users] Ways of Integrating Sphinx...
BTW Brian, Here is something that would make FS's VAD much better. The technique also improved Sphinx-3 performance in low-SNR enviroments and made it run over 40% faster. http://figment.cse.usf.edu/~sfefilat/data/papers/WeBT5.3.pdf Mark. -Original Message- From: Brian West br...@freeswitch.org To: freeswitch-users@lists.freeswitch.org Sent: Sat, 2 May 2009 7:42 am Subject: Re: [Freeswitch-users] Ways of Integrating Sphinx... On May 1, 2009, at 6:03 PM, mszla...@aol.com wrote: Hi Moiz, I've checking out mod_pocketshinx against other asr's on Windows with the same hardware.? No matter what settings one tries, mod_pocketsphinx is virtually unusable in real world scenarios.? I have used it and it works fine... I think your expectations are a bit high for it... Complex things like dictation is not what PocketSphinx is for. ?You should try linux cuz I know it works great there. One can play around with mod_pocketsphinx settings so that it picks voice up well but then there better not be any background noise either from a bad connection or just everyday sounds.? There is no other ASR out there that doesn't get pissed off at background noise or any noise for that matter... have you called ATT and Sprint lately? ?My dogs barking in the background really send theirs into fits and they paid tons of money for it. ? It just way to sensitive and of couse you'll notice this problem most with cell phones. Same with commercial ASR, Granted the acoustical model for PocketSphinx wasn't done with any files recorded from cellphone from what I can tell. ?You can do adaptation of the acoustical model as per the Sphinx wiki to make it more accurate for your needs that takes time and effort but it works. If you adjust the settings to try blocking out background noise you'll find you don't suceed all that well and then there are problems picking up the callers voice. Those settings are for telling when the person stopped talking... nothing more. It looks like some kind of signal pre-processing is required that isn't in place yet but we all know that this is a work-in progress. I'm not working on it... I run the pizza demo with PS and it works from my polycom rather well I would say it gets some things wrong but it does score them low so you can verify it in your scripts. I don't know if esl would make any difference. To use FS and an ASR/TTS I just bridge calls to another ASR application for now.? Mark Brian West br...@freeswitch.org -- Meet us at ClueCon! ?http://www.cluecon.com = ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Ways of Integrating Sphinx...
Nope it isn't but does that make a difference if pocketsphinx could use a similar upgrade? Anyway, you now have a way to make VAD better in FS. -Original Message- From: Brian West br...@freeswitch.org To: freeswitch-users@lists.freeswitch.org Sent: Mon, 4 May 2009 10:29 am Subject: Re: [Freeswitch-users] Ways of Integrating Sphinx... Wasn't aware Sphinx 3 was integrated into FreeSWITCH ...? /b On May 4, 2009, at 12:00 PM, mszla...@aol.com wrote: BTW Brian, Here is something that would make FS's VAD much better. The technique also improved Sphinx-3 performance in low-SNR enviroments and made it run over 40% faster. http://figment.cse.usf.edu/~sfefilat/data/papers/WeBT5.3.pdf Mark. Brian West br...@freeswitch.org -- Meet us at ClueCon! ?http://www.cluecon.com = ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] Latest SVN update gives Windows Express compiler errors ...
I'm getting Windows Express compiler errors on the latest svn update to trunk 13213. It looks like the path is wrong to some files. Instead of folder Debug, it's looking for files in folder Debug DLL 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
Re: [Freeswitch-users] FS 13157 fsctl shutdown restart sequence fails ...
It's running in the foreground on a 2.66 Ghz dual core with? ~3.5 G ram. OS is Windows XP SP3. -Original Message- From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Mon, 27 Apr 2009 7:22 am Subject: Re: [Freeswitch-users] FS 13157 fsctl shutdown restart sequence fails ... Are you running it in the foreground so you can see what happens? I do that many times a day and i have no seen it. On Mon, Apr 27, 2009 at 7:31 AM, Karl Vesterling k...@ken-ton.com wrote: I've been noticing that for a while now, probably the last 3 weeks. It's more frequent on slower systems (2.2Ghz and below), and ALWAYS the case on one dual-core 800Mhz. (two CPU's) Best Regards, Karl J. Vesterling k...@ken-ton.com 202-461-3231 x0 On Apr 26, 2009, at 11:58 PM, mszla...@aol.com wrote: In FS trunk 13157, fsctl shutdown restart fails to work as in the past. I get the complete shutdown without the restart. Mark. Green cleaning products -- do they work as well? Find out 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@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 ___ 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] FS 13157 fsctl shutdown restart sequence fails ...
I've been using fsctl shutdown restart with as far back as FS 12653M and probably earlier in the exact same manner from the same directory as FreeSWITCH.exe, i.e. \FreeSWITCH\Debug\ I think it ends with something like stop memory pooling but the window is gone to fast for me to be sure. -Original Message- From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Mon, 27 Apr 2009 9:30 am Subject: Re: [Freeswitch-users] FS 13157 fsctl shutdown restart sequence fails ... did that actually work in windows, i don't think it's been tested there. does it print anything at the end? is your CWD the same as in are you starting it from a cmd.exe in the same dir as the binary or setting the working dir to that place? On Mon, Apr 27, 2009 at 10:37 AM, mszla...@aol.com wrote: It's running in the foreground on a 2.66 Ghz dual core with? ~3.5 G ram. OS is Windows XP SP3. -Original Message- From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Mon, 27 Apr 2009 7:22 am Subject: Re: [Freeswitch-users] FS 13157 fsctl shutdown restart sequence fails ... Are you running it in the foreground so you can see what happens? I do that many times a day and i have no seen it. On Mon, Apr 27, 2009 at 7:31 AM, Karl Vesterling k...@ken-ton.com wrote: I've been noticing that for a while now, probably the last 3 weeks. It's more frequent on slower systems (2.2Ghz and below), and ALWAYS the case on one dual-core 800Mhz. (two CPU's) Best Regards, Karl J. Vesterling k...@ken-ton.com 202-461-3231 x0 On Apr 26, 2009, at 11:58 PM, mszla...@aol.com wrote: In FS trunk 13157, fsctl shutdown restart fails to work as in the past. I get the complete shutdown without the restart. Mark. Green cleaning products -- do they work as well? Find out 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@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 Can't afford a new spring wardrobe? Go shopping in your closet instead! ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@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 ___ 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] FS 13157 fsctl shutdown restart sequence fails ...
Oops, forgot something. If I do it by setting the working directory from cmd and then running FS from there then the fsctl shutdown restart does work. -Original Message- From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Mon, 27 Apr 2009 9:30 am Subject: Re: [Freeswitch-users] FS 13157 fsctl shutdown restart sequence fails ... did that actually work in windows, i don't think it's been tested there. does it print anything at the end? is your CWD the same as in are you starting it from a cmd.exe in the same dir as the binary or setting the working dir to that place? On Mon, Apr 27, 2009 at 10:37 AM, mszla...@aol.com wrote: It's running in the foreground on a 2.66 Ghz dual core with? ~3.5 G ram. OS is Windows XP SP3. -Original Message- From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Mon, 27 Apr 2009 7:22 am Subject: Re: [Freeswitch-users] FS 13157 fsctl shutdown restart sequence fails ... Are you running it in the foreground so you can see what happens? I do that many times a day and i have no seen it. On Mon, Apr 27, 2009 at 7:31 AM, Karl Vesterling k...@ken-ton.com wrote: I've been noticing that for a while now, probably the last 3 weeks. It's more frequent on slower systems (2.2Ghz and below), and ALWAYS the case on one dual-core 800Mhz. (two CPU's) Best Regards, Karl J. Vesterling k...@ken-ton.com 202-461-3231 x0 On Apr 26, 2009, at 11:58 PM, mszla...@aol.com wrote: In FS trunk 13157, fsctl shutdown restart fails to work as in the past. I get the complete shutdown without the restart. Mark. Green cleaning products -- do they work as well? Find out 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@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 Can't afford a new spring wardrobe? Go shopping in your closet instead! ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@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 ___ 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] FS 13157 fsctl shutdown restart sequence fails ...
See my addendum (last post). This way it works and does restart. -Original Message- From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Mon, 27 Apr 2009 11:41 am Subject: Re: [Freeswitch-users] FS 13157 fsctl shutdown restart sequence fails ... open the cli.exe manually first and cd to the FS dir and run it from there so when it exits it will drop back to the prompt and you can read it On Mon, Apr 27, 2009 at 12:51 PM, mszla...@aol.com wrote: I've been using fsctl shutdown restart with as far back as FS 12653M and probably earlier in the exact same manner from the same directory as FreeSWITCH.exe, i.e. \FreeSWITCH\Debug\ I think it ends with something like stop memory pooling but the window is gone to fast for me to be sure. -Original Message- From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Mon, 27 Apr 2009 9:30 am Subject: Re: [Freeswitch-users] FS 13157 fsctl shutdown restart sequence fails ... did that actually work in windows, i don't think it's been tested there. does it print anything at the end? is your CWD the same as in are you starting it from a cmd.exe in the same dir as the binary or setting the working dir to that place? On Mon, Apr 27, 2009 at 10:37 AM, mszla...@aol.com wrote: It's running in the foreground on a 2.66 Ghz dual core with? ~3.5 G ram. OS is Windows XP SP3. -Original Message- From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Mon, 27 Apr 2009 7:22 am Subject: Re: [Freeswitch-users] FS 13157 fsctl shutdown restart sequence fails ... Are you running it in the foreground so you can see what happens? I do that many times a day and i have no seen it. On Mon, Apr 27, 2009 at 7:31 AM, Karl Vesterling k...@ken-ton.com wrote: I've been noticing that for a while now, probably the last 3 weeks. It's more frequent on slower systems (2.2Ghz and below), and ALWAYS the case on one dual-core 800Mhz. (two CPU's) Best Regards, Karl J. Vesterling k...@ken-ton.com 202-461-3231 x0 On Apr 26, 2009, at 11:58 PM, mszla...@aol.com wrote: In FS trunk 13157, fsctl shutdown restart fails to work as in the past. I get the complete shutdown without the restart. Mark. Green cleaning products -- do they work as well? Find out 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@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 Can't afford a new spring wardrobe? Go shopping in your closet instead! ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:213-799-1400 ___ Freeswitch-users mailing list
[Freeswitch-users] Problem with a hang up event on transfer.
Using 13157 and some prior builds, I'm having a problem with the hang up hook activating on a transfer at the end of the script. The hook is suppose to fire with hang ups except when the transfer occurs and I thought setAutoHangup(false) would prevent this but it doesn't. Here is how things are set up. session.setHangupHook(on_hangup); session.setAutoHangup(false); session.execute(transfer, NOTIFY_ME XML default); session = undefined; If setAutoHangup(false) is working correctly then is there a method to unset the hang up hook before the transfer? 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
Re: [Freeswitch-users] Problem with a hang up event on transfer.
Got it. Thanks Tony. -Original Message- From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Mon, 27 Apr 2009 1:59 pm Subject: Re: [Freeswitch-users] Problem with a hang up event on transfer. it happens on both hangup and transfer you look at the arg to the hangup hook to tell if it was transfer or hangup and act accordingly On Mon, Apr 27, 2009 at 3:40 PM, Brian West br...@freeswitch.org wrote: This only makes the call not hangup when the script exits. /b On Apr 27, 2009, at 3:29 PM, mszla...@aol.com wrote: If setAutoHangup(false) is working correctly then is there a method to unset the hang up hook before the transfer? Mark. Brian West br...@freeswitch.org -- Meet us at ClueCon! ?http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@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 ___ 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] Compact, fanless appliance?
1. Hell no. -Original Message- From: Fred-145 codecompl...@free.fr To: freeswitch-users@lists.freeswitch.org Sent: Fri, 24 Apr 2009 3:30 am Subject: Re: [Freeswitch-users] Compact, fanless appliance? Fred-145 wrote: I think it'd be very useful if the wiki included a list of devices that can run Freeswitch in addition to a regular PC, especially those that cost $150. Here's a list of makers of basic routers per www.cheaprouter.us: 2Wire, 3Com, ADTRAN, Apple, Belkin, CNet, CP Technology, Cisco, Cisco Systems, Compaq, D-Link, Edimax, Efficient Networks, Hawking Technology, Hewlett-Packard, Linksys, Motorola, Netgear, Netopia, Nortel, SMC, TP-Link, TRENDnet, Zoom, ZyXel For those of you used to tinker with this type of basic hardware... 1. Can a non-expert go from opening the box to getting a Freeswitch server up and running without spending days figuring it out? 2. Are they fast enough for SOHO use (a couple of simultaneous SIP conversations + small LAMP web server)? 3. Do they offer a CompactFlash plug to provide a few gigabytes of storage before plugging an external drive? Thank you. -- View this message in context: http://www.nabble.com/Compact%2C-fanless-appliance--tp23193738p23213781.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
Re: [Freeswitch-users] VoiceXML
This was beyond what I was thinking about. I was plenty happy with just writing the vxml tags. But if this works well then go for it. -Original Message- From: Remko Kloosterman r.klooster...@mtel.nl To: freeswitch-users@lists.freeswitch.org Sent: Fri, 24 Apr 2009 12:31 pm Subject: Re: [Freeswitch-users] VoiceXML Excellent! I wasn't aware of?their development progress. Let's check it out. Van: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] Namens EdPimentl Verzonden: vrijdag 24 april 2009 15:10 Aan: freeswitch-users@lists.freeswitch.org Onderwerp: Re: [Freeswitch-users] VoiceXML Here is one editor to look into http://www.eclipse.org/vtp/ -E Gpro.ws http://TwiTR.Me ? ? ? ? ?(Cross Social Network Messaging Bus) http://WatchNtweet.Me (Watch and Chat/Tweet) SocialTV http://TwebEX.com ? ? (Twitter Based Online Web Conference Platform) http://TweetUp.ws ? ? ?(Twitter based ?MeetUp service) ___ 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] VoiceXML
If you don't like vxml then here is a post on voicePHP http://www.speechtechblog.com/2009/04/22/voicexml-to-go-down-in-the-third-says-voicephp It's from a vendor but there might be some good ideas to get from what they are doing. FreeSWITCH needs demand to get vxml and it's not there yet. For now, it looks like the FS community is waiting for demand instead of trying to create it. David Knell wrote: On Tue, 2009-04-21 at 14:35 -0400, mszla...@aol.com wrote: Great Idea. Try setting up the exact same dialogue with say Voxeo's VoiceXML system and then with Javascript/Lua and pocketsphinx. It's an order of magnitude faster with VoiceXML. Out of interest, is that using some RAD tool or coding directly in VoiceXML? I ask because VoiceXML strikes me as being a bastard abomination of the highest order, whose sole saving grace is that it's a standardised bastard abomination. Or is Pocketsphinx the problem? Cheers -- Dave ___ 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 -- View this message in context: http://www.nabble.com/VoiceXML-tp23161671p23198458.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
Re: [Freeswitch-users] VoiceXML
Well, far to many times awesome things died because there was nothing that created the demand. After all, build it and they will come just gets business people rolling their eyes. Ask yourself how much crap gets sold just because of creating demand while the good stuff struggles because no demand is being created. Happens all the time from cloths to entertainment to tech to health care. Now, you maybe right that vxml isn't awesome but that wasn't my experience. -Original Message- From: Michael Collins m...@freeswitch.org To: freeswitch-users@lists.freeswitch.org Sent: Thu, 23 Apr 2009 1:09 pm Subject: Re: [Freeswitch-users] VoiceXML On Thu, Apr 23, 2009 at 12:49 PM, mszlazak mszla...@aol.com wrote: If you don't like vxml then here is a post on voicePHP http://www.speechtechblog.com/2009/04/22/voicexml-to-go-down-in-the-third-says-voicephp It's from a vendor but there might be some good ideas to get from what they are doing. Interesting, except for the PHP part. ;) ? FreeSWITCH needs demand to get vxml and it's not there yet. For now, it looks like the FS community is waiting for demand instead of trying to create it. Correct. VXML does need us to create demand. If it is as awesome as some would have us believe then the market will drive the demand. -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ 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] VoiceXML
Well if it's for marketing reasons then maybe that's why FS should do it. After all, the idea is to get people to try something. Plus it is fast in making up a dialogue so someone can quickly test if this will work for what they want. In a previous post, Steve said many don't use it but Voxeo Forums do have posts daily on vxml. Maybe it's the first-timers getting their feet wet. David Knell wrote: Just sticking 'voice' in front of something doesn't automatically make it a good tool for developing voice applications - there's more marketing here than anything else. And it's not like adding extensions to an existing language to provide IVR control is anything new: it's exactly what you get if you develop for FS in Javascript, Lua or any of its other supported languages. From my point of view, as a programmer, VoiceXML is the wrong idiom for development of IVR/telephony services; a procedural language works just fine. I suspect that I'm not alone, and I further suspect that that's why there's no real push to get VoiceXML supported. --Dave If you don't like vxml then here is a post on voicePHP http://www.speechtechblog.com/2009/04/22/voicexml-to-go-down-in-the-third-says-voicephp It's from a vendor but there might be some good ideas to get from what they are doing. FreeSWITCH needs demand to get vxml and it's not there yet. For now, it looks like the FS community is waiting for demand instead of trying to create it. David Knell wrote: On Tue, 2009-04-21 at 14:35 -0400, mszla...@aol.com wrote: Great Idea. Try setting up the exact same dialogue with say Voxeo's VoiceXML system and then with Javascript/Lua and pocketsphinx. It's an order of magnitude faster with VoiceXML. Out of interest, is that using some RAD tool or coding directly in VoiceXML? I ask because VoiceXML strikes me as being a bastard abomination of the highest order, whose sole saving grace is that it's a standardised bastard abomination. Or is Pocketsphinx the problem? Cheers -- Dave ___ 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 -- View this message in context: http://www.nabble.com/VoiceXML-tp23161671p23208811.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
Re: [Freeswitch-users] VoiceXML
I've used vxml on Voxeo's system and it's really nice to work with. Underneath it's tags is Javascript so a FreeSWITCH with a fast TraceMonkey engine and vxml would be great. Your MRCP project would help connect things to other ASR/TTS systems if pocketsphinx isn't good enough. Nice package. -Original Message- From: Arsen Chaloyan achalo...@yahoo.com To: freeswitch-users@lists.freeswitch.org Sent: Wed, 22 Apr 2009 3:15 am Subject: Re: [Freeswitch-users] VoiceXML VXML is not only interesting but required concept. I'm not talking about VXML itself, but presence of specification. I still remember my confusion, when initially thinking about media and call control dialogs and interfaces I found out XML based languages such as VXML and CCXML. Nevertheless VXML clearly defines media dialogs: entities, state machines and transitions, which is just good. More over VXML 3.0 is under the way http://www.w3.org/TR/vxml30reqs/ If you check the authors list, it will be clear this is going to be another standart we should live with. 3.0 is rework of 2.1 version, added native support of audio/video dialogs, speaker identification/verification. FreeSWITCH as a platform is almost ready for that, what it is still missing is VXML interpreter itself. Regards, Arsen. www.unimrcp.org Date: Wed, 22 Apr 2009 09:41:50 +0200 From: Remko Kloosterman r.klooster...@mtel.nl Subject: Re: [Freeswitch-users] VoiceXML To: freeswitch-users@lists.freeswitch.org Message-ID: ??? 11372c8b9e603f4facde6ab18256dec60170f...@srvmtel.office.mtel.nl Content-Type: text/plain; charset=us-ascii I agree that the original commercial model used for VXML gets in the way of it's own success. It's also focussed way too much on ASR/TTS. I think we can all agree this technology is still a future promise, even after 10+ years. But technically VXML is an interesting concept, especially together with related standards like ccxml (http://en.wikipedia.org/wiki/Call_Control_eXtensible_Markup_Language). The latter supports call transfer and conferencing. Every VoIP and TDM product I've worked with in the past has it's own application interface or is bound to an interface that's tailored for a special use (like pbx). I've also used voxeo's callxml, that's implemented with the CosmoCom ACD product. It's implementation is crappy, but the concept is nice. The idea of separating callcontrol+media (voice/swithing core) and application+ivr logic (vxml webserver cgi scripts) sounds rather appealing to me. Do you known the name of the free VXML editor/client that's out there? Van: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] Namens Anthony Minessale Verzonden: woensdag 22 april 2009 3:58 Aan: freeswitch-users@lists.freeswitch.org Onderwerp: Re: [Freeswitch-users] VoiceXML I was initially turned off by VXML when it came out because the first way they tried to make money off it was to sell the clients that let you build the actual xml.? I was not really motivated to pay money to be able to just generate xml just so i could code a free server for it so I lost interest. I did hear there is now finally a free one out there. so that may make it a little more reasonable. I've commented in the past that I'm totally open to supporting VXML but we have never had the public interest, time or resources to work on it thus far. On Tue, Apr 21, 2009 at 6:15 PM, David Knell d...@3c.co.uk wrote: ??? On Tue, 2009-04-21 at 14:35 -0400, mszla...@aol.com wrote: ??? Great Idea. ??? Try setting up the exact same dialogue with say Voxeo's VoiceXML ??? system and then with Javascript/Lua and pocketsphinx. It's an order of ??? magnitude faster with VoiceXML. ??? ??? ??? Out of interest, is that using some RAD tool or coding directly in ??? VoiceXML?? I ask because VoiceXML strikes me as being a bastard ??? abomination of the highest order, whose sole saving grace is that ??? it's a standardised bastard abomination. ??? ??? Or is Pocketsphinx the problem? ??? ??? Cheers -- ??? ??? Dave ??? ??? ___ ??? Freeswitch-users mailing list ??? Freeswitch-users@lists.freeswitch.org ??? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users ??? UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users ??? http://www.freeswitch.org ??? -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org
Re: [Freeswitch-users] FXO cards for freeswitch on windows
Try searching or getting help from this forum. http://forum.voxilla.com/linksys-sipura-voip-support-forum/ Look to hwittenb for further assistance. His customer service skills are excellent. -Original Message- From: xbipin bi...@xbipin.com To: freeswitch-users@lists.freeswitch.org Sent: Wed, 22 Apr 2009 8:16 am Subject: [Freeswitch-users] FXO cards for freeswitch on windows i was just thinking of throwing away my spa3102 and setting my windows machine with FS on it to conenct to the telephone line for incomming calls, something like a calling card gateway using a FXO card. the reason being from the time i bought the spa3102, i haven't been able to get the caller id on it to work with the specs of my country and the linksys ppl also have no clue about it nor do they provide any help on it so im just sick and tired of it so i wanted to know if any single port FXO cards work with FS or no, if so which ones and that too on a windows machine. and secondly if any1 of the sellers on this mailing list could ship it to my location coz its not available in my country. i read somewhere that a normal 56k dialup modem can be sued like a FXO card, any ideas on it? -- View this message in context: http://www.nabble.com/FXO-cards-for-freeswitch-on-windows-tp23175519p23175519.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
Re: [Freeswitch-users] VoiceXML
Great Idea. Try setting up the exact same dialogue with say Voxeo's VoiceXML system and then with Javascript/Lua and pocketsphinx. It's an order of magnitude faster with VoiceXML. I don't know went demand will pick up for VXML on FS but one issue that I think will block it is how robust the FS embedding of pocketsphinx is to the harsh noisy enviroment of telephony. Currently, it lacks this robustness compared to vendor products but once that happens then more will want to use it and VXML will become more in demand? ... hopefully. -Original Message- From: Remko Kloosterman r.klooster...@mtel.nl To: freeswitch-users@lists.freeswitch.org Sent: Tue, 21 Apr 2009 10:57 am Subject: [Freeswitch-users] VoiceXML Hi everyone, ? Did someone ever try and implement a VXML interface on freeswitch? Or do you think it's a good idea? Or not? Since it is an actual standard, I guess there might be a market for application service providers. ? Remko ___ 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] VoiceXML
Directly in vxml. -Original Message- From: David Knell d...@3c.co.uk To: freeswitch-users@lists.freeswitch.org Sent: Tue, 21 Apr 2009 4:15 pm Subject: Re: [Freeswitch-users] VoiceXML On Tue, 2009-04-21 at 14:35 -0400, mszla...@aol.com wrote: Great Idea. Try setting up the exact same dialogue with say Voxeo's VoiceXML system and then with Javascript/Lua and pocketsphinx. It's an order of magnitude faster with VoiceXML. Out of interest, is that using some RAD tool or coding directly in VoiceXML? I ask because VoiceXML strikes me as being a bastard abomination of the highest order, whose sole saving grace is that it's a standardised bastard abomination. Or is Pocketsphinx the problem? Cheers -- Dave ___ 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] Ideas for my presentation
Start by asking Tony what pain he was experiencing that made him create FreeSWITCH in the first place. What's posted about FS on Wikipedia does not go into it that much. I'm not an IT guy but I can _imagine_ how life _might_ be easier with FS. *Using FS requires less maintenance, like it crashes less. So on my days off I don't get as many calls to come to the office. I have more time with friends and family since using FS. There are less complaints from my supervisors and customers using the service and costs the company less, etc. * Get into more specific pains like,? When it does crash it's quicker and easier to fix ... * When it runs, it runs well because ... * I can upgrade and extend it easily ... I'm assuming that you pretty much got a handle on developer needs, but you get the idea. Very general pain issues about their lives then get more specific stuff. On the vision thing, well I doubt FS is the future of telephony but that's because I doubt that voip is the future of telephony. -Original Message-From: Diego Viola diego.vi...@gmail.com To: mszla...@aol.com Sent: Mon, 20 Apr 2009 4:40 am Subject: Re: [Freeswitch-users] Ideas for my presentation Hey guys, Any ideas on what I could show for the demo? I would like to show something really impressive that will make people go omg, wow. I was thinking in showing a stress testing with sipp or something, and a real call on the background... to show how fast FS is, or to show how much load can handle. But I think I could just say that... I'd like to do something more impressive, any ideas? Diego On Sun, Apr 19, 2009 at 11:55 PM, Diego Viola diego.vi...@gmail.com wrote: Yep, I'm also separating now everything that is FreeSWITCH and the business side into different sections. So I can talk about FreeSWITCH entirely first and then the business. Thanks everyone :) Diego On Sun, Apr 19, 2009 at 11:43 PM, mszla...@aol.com wrote: The slides look nice but it's hard for me to judge much without their accompanying dialog. -Original Message- From: Diego Viola diego.vi...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Sun, 19 Apr 2009 5:46 pm Subject: Re: [Freeswitch-users] Ideas for my presentation Ok, here is my current draft for the presentation. http://voip.richapplabs.com/freeswitch/freeswitch.pdf Enjoy :D PS: If you have any suggestions or feedback keep them coming, long life FreeSWITCH! Thanks, Diego On Sun, Apr 19, 2009 at 3:31 PM, bakko asannu...@gmail.com wrote: Me too :) ? we want slides! we want slides! we want slides! ___ 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 Why pay full price? Check out this month's deals on the new AOL Shopping. ___ 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] Ideas for my presentation
Start of with the big picture. What is FreeSWITCH? Tony talked about it in an interview so you have his thoughts. I see it as an audio/speech real-time communications platform for many OS's that connects multiple users independent of their audio format with a current focus on telephony. Maybe it could be extended to other areas of audio as well. It scales from an abacus or iphone all the way to a supercomputer. Like my cleaning lady, it does Windows. ? Anyway, start of with an elevator speech then go from there. -Original Message- From: Diego Viola diego.vi...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Sat, 18 Apr 2009 3:28 am Subject: [Freeswitch-users] Ideas for my presentation Hi guys, I'm going to give a presentation of FreeSWITCH for our next Flisol ( http://www.flisol.net/ ) the next Saturday 25. I currently have some cool ideas of what to say and what to show to them, but I'm looking for more, in case that you have it. My audience will be mostly people interested in the technology and as well as business. Let me know if you have some nice ideas for my presentation, I already got some by myself, but more are always welcome :). Thanks, Diego ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Ideas for my presentation
BTW, here is a video on how to craft an elevator pitch/speech in an investor or V.C. context, the rules still mostly apply in other areas. It's becomes the into to all that follows. http://www.youtube.com/watch?v=Tq0tan49rmc Good luck in the talk. -Original Message- From: mszla...@aol.com To: freeswitch-users@lists.freeswitch.org Sent: Sat, 18 Apr 2009 8:30 am Subject: Re: [Freeswitch-users] Ideas for my presentation Start of with the big picture. What is FreeSWITCH? Tony talked about it in an interview so you have his thoughts. I see it as an audio/speech real-time communications platform for many OS's that connects multiple users independent of their audio format with a current focus on telephony. Maybe it could be extended to other areas of audio as well. It scales from an abacus or iphone all the way to a supercomputer. Like my cleaning lady, it does Windows. ? Anyway, start of with an elevator speech then go from there. -Original Message- From: Diego Viola diego.vi...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Sat, 18 Apr 2009 3:28 am Subject: [Freeswitch-users] Ideas for my presentation Hi guys, I'm going to give a presentation of FreeSWITCH for our next Flisol ( http://www.flisol.net/ ) the next Saturday 25. I currently have some cool ideas of what to say and what to show to them, but I'm looking for more, in case that you have it. My audience will be mostly people interested in the technology and as well as business. Let me know if you have some nice ideas for my presentation, I already got some by myself, but more are always welcome :). Thanks, Diego ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Why pay full price? Check out this month's deals on the new AOL Shopping. ___ 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] Problem with originate in javascript.
Hi Tony,
But I thought we settled on Janitor. ;-)
BTW, it was the other point about keeping the FS founders involved or not in
the documentation process that concerned much much more. That was the big issue
that got me going on that thread.
Anyway, I appreciate your help and will do some document engineering but need
one further elaboration on de-referencing the original session object since I
tried the execute(transfer ...) before and couldn't get that to work. Can you
show me an example and I can then put up both approaches.
Mark.
-Original Message-
From: Anthony Minessale anthony.miness...@gmail.com
To: freeswitch-users@lists.freeswitch.org
Sent: Thu, 9 Apr 2009 6:09 am
Subject: Re: [Freeswitch-users] Problem with originate in javascript.
The 2nd 2 examples you provided are invalid, they depict the usage of the
originate api command in the context of the constructor
to a JS session.
If you want to send the call to another extension you have to create the
channel like you did in the first example followed by
session.execute(transfer, GINO_ANS XML default);
at which time it would be wise if you deref the session object because its
thread will be running in the new extension.
A better way would be to do both in one with a single call to the originate api
command
apiExecute(originate,
{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/14082031...@10.0.0.5:5061
GINO_ANS XML default);
This never gives you a session object it just creates a channel and transfers
it to the desired extension.
A Documentation Re-factorial Engineer may be able to add it to the relevant
page on the wiki if it is not already present.
On Wed, Apr 8, 2009 at 6:15 PM, mszla...@aol.com wrote:
I want to run a script with a scheduler but I'm having a problem with how to
set up the originate in Javascript.
The originate would go something like:
originate
{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/12223334...@10.0.0.5:5061
GINO_ANS
I can get this to work:
session = new
Session({id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/14082031...@10.0.0.5:5061);
But I want to drop that into an extension that runs another script and can't
get either of these to work:
session = new
Session({id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/14082031...@10.0.0.5:5061
GINO_ANS);
session = new
Session({id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/14082031...@10.0.0.5:5061
GINO_ANS XML default);
Also, will I have problems running the second script from the first script?
Thanks.
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Re: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption
I'm not quite sure what your asking.
Are you saying that I could run the latest FS svn but in a way that uses my
older configuration files? If so then I don't, and don't know how ... blush
blush.
If that's the easiest thing to do then please tell me how.
Thanks. Mark.
-Original Message-
From: Szymon Olko so...@gcdf.pl
To: freeswitch-users@lists.freeswitch.org
Sent: Tue, 7 Apr 2009 10:54 pm
Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1
and memory consumption
mszla...@aol.com pisze:
I have to change $${local_ip_v4} to 10.0.0.3 every time I update to a
new trunk and I have to go through vars.xml, etc changing
$${local_ip_v4} like you did.
Is there a way to change $${local_ip_v4} in one place. That way one
wouldn't have remember all the locations that it needs to be changed?
My configuration is not updated when I compile new version and install it. Do
you run FS with configuration path pointed to svn
work dir?
-Original Message-
From: Peter P GMX prometheus...@gmx.net
To: freeswitch-users@lists.freeswitch.org
Sent: Tue, 7 Apr 2009 3:52 pm
Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on
127.0.0.1 and memory consumption
Thanks Brian,
what I was actually looking for was to use a standard SIP soft phone
with some additional features.
I finally manged to make FS listen on 127.0.0.1 the following way:
vars.xml
X-PRE-PROCESS cmd=set data=domain=127.0.0.1/
internal.xml
param name=rtp-ip value=127.0.0.1/
param name=sip-ip value=127.0.0.1/
The rest is standard configuration.
Now communication Laptop-internal is UDP on port 5060 and external via
TLS on port 5081, so I have no open port 5060 to the internet.
Best regards
Peter
Brian West schrieb:
On Apr 7, 2009, at 12:19 PM, Peter P GMX wrote:
1st Question: Is that possible or is another solution preferrable?
Just use FreeSWITCH with mod_portaudio.
2nd Question: How can I change the amount of memory FS tries to reserve
to an absolute minumum (I only have 1 call at a time). Currently it
tries to reserve about 360M if I read that right.
Thats virtual. Look at RES.
Best regards
Peter
Brian West
br...@freeswitch.org mailto:br...@freeswitch.org
mailto:br...@freeswitch.org
mailto:br...@freeswitch.org?
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Re: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption
Na jakiś czas, będę uczyć się, jak radzić sobie z Tortoise SVN błędów.
Mój polski nie jest zbyt dobre, ale dziękuję. Google pomaga.
-Original Message-
From: Szymon Olko so...@gcdf.pl
To: freeswitch-users@lists.freeswitch.org
Sent: Wed, 8 Apr 2009 10:36 am
Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1
and memory consumption
mszla...@aol.com pisze:
OK , you're SVN updating on a Linux system but I'm using Windows. The
very few times I tried with Tortoise SVN I ran into problems were it
would fail because of some path not being present or some strange symbol
in a file or something else. Since I'm not experienced enough and don't
always have the time, I gave up on this approach and just start over
again in a different folder then reconfigure the updated FS and transfer
files from an older FS. Yup, it sucks.
Yes I'm linux user.
If you have problems with svn update then you can do your way, make fresh
checkout every time. After CO copy modules.conf and
build new version, just copy old config files to installation directory if it
is
always different one.
That's why I hate gui tools for things like full svn update.
-Original Message-
From: Szymon Olko so...@gcdf.pl
To: freeswitch-users@lists.freeswitch.org
Sent: Wed, 8 Apr 2009 12:59 am
Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on
127.0.0.1 and memory consumption
=0
A mszla...@aol.com mailto:mszla...@aol.com pisze:
I'm not quite sure what your asking.
Are you saying that I could run the latest FS svn but in a way that uses
my older configuration files? If so then I don't, and don't know how
... blush blush.
If that's the easiest thing to do then please tell me how.
Thanks. Mark.
Exactly, I do it that way.
For first time I gave installation prefix when configuring FS. You can stay
with
/usr/local/freeswich/.
Now every time i call 'make current' and it does not overwrite my
configuration
file.
In case of huge changes in modules I copy/merge my config file with the one
from
svn. I did not have problems with it, because
developers makes good default values for new configuration options.
I don't know which modules do you use, but in ones I use configuration is not
changes a lot, there are new options added which
does not break old one.
make install do not copy configuration files for me if they are already
installed, I have that on production server and all test
servers. I assume this is correct behavior and I'm not the only one work like
that.
I looked in Makefile and it tests for config file before installing, so it
does
not overwrite them.
Regards
Szymon
=2
0
-Original Message-
From: Szymon Olko so...@gcdf.pl mailto:so...@gcdf.pl
To: freeswitch-users@lists.freeswitch.org
mailto:freeswitch-users@lists.freeswitch.org
Sent: Tue, 7 Apr 2009 10:54 pm
Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on
127.0.0.1 and memory consumption
mszla...@aol.com mailto:mszla...@aol.com mailto:mszla...@aol.com
mailto:mszla...@aol.com? pisze:
I have to change $${local_ip_v4} to 10.0.0.3 every time I update to a
new trunk and I have to go through vars.xml, etc changing
$${local_ip_v4} like you did.
Is there a way to change $${local_ip_v4} in one place. That way one
wouldn't have remember all the locations that it needs to be changed?
My configuration is not updated when I compile new version and install it.
Do
you run FS with configuration path pointed to svn
work dir?
-Original Message-
From: Peter P GMX prometheus...@gmx.net mailto:prometheus...@gmx.net
mailto:prometheus...@gmx.net mailto:prometheus...@gmx.net?
To: freeswitch-users@lists.freeswitch.org
mailto:freeswitch-users@lists.freeswitch.org
mailto:freeswitch-users@lists.freeswitch.org
mailto:freeswitch-us...@lists.freeswitch.org?
Sent: Tue, 7 Apr 2009 3:52 pm
Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on
127.0.0.1 and memory consumption
Thanks Brian,
what I was actually looking for was to use a standard SIP soft phone
with some additional features.
I finally manged to make FS listen on 127.0.0.1 the following way:
vars.xml
X-PRE-PROCESS cmd=set data=domain=127.0.0.1/
internal.xml
param name=rtp-ip value=127.0.0.1/
param name=sip-ip value=127.0.0.1/
The rest is standard configuration.
Now communication Laptop-internal is UDP on port 5060 and external via
TLS on port 5081, so I have no open port 5060 to the internet.
[Freeswitch-users] Problem with originate in javascript.
I want to run a script with a scheduler but I'm having a problem with how to
set up the originate in Javascript.
The originate would go something like:
originate
{id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/12223334...@10.0.0.5:5061
GINO_ANS
I can get this to work:
session = new
Session({id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/14082031...@10.0.0.5:5061);
But I want to drop that into an extension that runs another script and can't
get either of these to work:
session = new
Session({id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/14082031...@10.0.0.5:5061
GINO_ANS);
session = new
Session({id_name='${caller_id_name}',id_number=${caller_id_number}}sofia/gateway/spa3102PSTN/14082031...@10.0.0.5:5061
GINO_ANS XML default);
Also, will I have problems running the second script from the first script?
Thanks.
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Re: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption
I have to change $${local_ip_v4} to 10.0.0.3 every time I update to a new
trunk and I have to go through vars.xml, etc changing $${local_ip_v4} like you
did.
Is there a way to change $${local_ip_v4} in one place. That way one wouldn't
have remember all the locations that it needs to be changed?
-Original Message-
From: Peter P GMX prometheus...@gmx.net
To: freeswitch-users@lists.freeswitch.org
Sent: Tue, 7 Apr 2009 3:52 pm
Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1
and memory consumption
Thanks Brian,
what I was actually looking for was to use a standard SIP soft phone
with some additional features.
I finally manged to make FS listen on 127.0.0.1 the following way:
vars.xml
X-PRE-PROCESS cmd=set data=domain=127.0.0.1/
internal.xml
param name=rtp-ip value=127.0.0.1/
param name=sip-ip value=127.0.0.1/
The rest is standard configuration.
Now communication Laptop-internal is UDP on port 5060 and external via
TLS on port 5081, so I have no open port 5060 to the internet.
Best regards
Peter
Brian West schrieb:
On Apr 7, 2009, at 12:19 PM, Peter P GMX wrote:
1st Question: Is that possible or is another solution preferrable?
Just use FreeSWITCH with mod_portaudio.
2nd Question: How can I change the amount of memory FS tries to reserve
to an absolute minumum (I only have 1 call at a time). Currently it
tries to reserve about 360M if I read that right.
Thats virtual. Look at RES.
Best regards
Peter
Brian West
br...@freeswitch.org mailto:br...@freeswitch.org
-- Meet us a ClueCon! http://www.cluecon.com http://www.cluecon.com/
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Re: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1 and memory consumption
Wonderful! Thank you sir.
-Original Message-
From: Jason White ja...@jasonjgw.net
To: freeswitch-users@lists.freeswitch.org
Sent: Tue, 7 Apr 2009 7:04 pm
Subject: Re: [Freeswitch-users] freeswitch on a laptop listening on 127.0.0.1
and memory consumption
mszla...@aol.com mszla...@aol.com wrote:
Is there a way to change $${local_ip_v4} in one place.
Of course. That's why it's a variable.
X-PREPROCESS cmd=set data=local_ip_v4=10.10.1.2/
this goes in vars.xml, substituting the desired address.
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[Freeswitch-users] MANDATORY_IE_MISSING and NORMAL_TEMPORARY_FAILURE using javascript in dialplan.
On Windows XP/SP3 with FS trunk 12653M I get these errors using javascript in
my dialplan:
?
[MANDATORY_IE_MISSING] (see pastebin below)
and/or with [CS_EXCHANGE_MEDIA] I get [NORMAL_TEMPORARY_FAILURE]? (not shown
this time in pastebin)
Here is the test javascript file:
session.answer();
session.setVariable(choice, demo);
If I remove session.answer() in the above test script then there is no problem
but that doesn't always work with another scripts.
Here is the related section of the dialplan. You dial into ext. and the
problem happens at action application=bridge
data=sofia/internal/${sip_to_us...@127.0.0.1:5068/. The variable choice is
set to demo to get here.
? ??? extension name=???
? ??? ??? condition field=destination_number expression=^$
? ??? ??? ??? !--action application=info/--
? ??? ??? ??? action application=set
data=effective_caller_id_name=${caller_id_name}/
? ??? ??? ??? action application=set
data=effective_caller_id_number=${caller_id_number}/
? ??? ??? ??? action application=set data=choice='not_saved'/
? ??? ??? ??? action application=javascript data=home_phone.js/? ??? ???
???
? ??? ??? ??? action application=transfer data=2223 XML default/
? ??? ??? /condition
? ??? /extension
? ???
??? ??? extension name=???
??? ??? ??? condition field=destination_number expression=^2223$/
? ??? ??? condition field=${choice} expression=^saved
? ??? ??? ??? action application=set data=api_hangup_hook=sched_api +60
none originate
{id_name='${caller_id_name}',id_number=${caller_id_number},id_message=${uuid}}sofia/gateway/spa3102PSTN/12223334...@10.0.0.5:5061
GINO_ANS/??? ??? ???
?? ??? ??? ??? anti-action application=transfer data=2224 XML default/
? ??? ??? /condition
? ??? /extension
? ???
? ??? extension name=???
? ??? ??? condition field=destination_number expression=^2224$/
?? ??? ??? condition field=${choice} expression=^demo
? ??? ??? ??? !-- Bridge to Voxeo Prophecy ASR/TTS --
? ??? ??? ??? action application=set
data=effective_caller_id_name=${caller_id_name}/
? ??? ??? ??? action application=set
data=effective_caller_id_number=${caller_id_number}/
?? ??? ??? ??? action application=set data=bypass_media=true/
??? ??? ??? action application=set data=hangup_after_bridge=true/
? ??? ??? ??? action application=bridge
data=sofia/internal/${sip_to_us...@127.0.0.1:5068/
? ??? ??? ??? anti-action application=hangup/
? ??? ??? /condition
? ??? /extension
? ??? extension name=
? ??? ??? condition field=destination_number expression=^GINO_ANS$
? ??? ??? ??? action application=answer/
? ??? ??? ??? action application=javascript data=notify_Gino.js
'${id_name}' ${id_number} ${id_message}/
? ??? ??? ??? action application=hangup/
? ??? ??? /condition
? ??? /extension
I enabled SIP/Sofia tracing and pastebinned part of the output here:
?http://pastebin.freeswitch.org/8321
Thanks.
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Re: [Freeswitch-users] MANDATORY_IE_MISSING and NORMAL_TEMPORARY_FAILURE using javascript in dialplan.
I svn'd to today's latest trunk to see if the problem remained. However,
things seemed to have turned worse. The error messages I had before don't
occur but I still can't bridge to my other application with choice=demo.
I tried the application by dialing straight into it with the following dial
plan. This has worked in FS 1.0.1 through 1.0.3 but fails to make any
connection in the current svn trunk.
NOTE ON UNRELATED ERROR: I don't use Lua but there was an error in compiling
LUA on Windows with 2008 Express so I get a error loading mod_lua.dll today.
Simple dialpan that worked before:
extension name=
condition field=destination_number expression=^2010$
!--action application=info/--
action application=set
data=effective_caller_id_name=${caller_id_name}/
action application=set
data=effective_caller_id_number=${caller_id_number}/
action application=set data=bypass_media=true/
!-- Bridge to Voxeo Prophecy ASR/TTS --
action application=bridge
data=sofia/internal/${sip_to_us...@127.0.0.1:5068/
/condition
/extension
SIP TRACE:
http://pastebin.freeswitch.org/8336
--
View this message in context:
http://www.nabble.com/MANDATORY_IE_MISSING-and-NORMAL_TEMPORARY_FAILURE-using-javascript-in-dialplan.-tp22912880p22918984.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.
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Re: [Freeswitch-users] Call For Help: Janitor Projects
First off. I would not call it a janitors project since that may offend some. A second problem is your notion that documentation is not-quite-as-important a task as writing code. I'm think many would say you have that backwards. There is nothing more effective in evolving FreeSwitch than good documentation which helps further development and is an important part of customer service. Good customer service is then a part of sales and marketing. Much more often than not, It's sales and marketing that is more important to making something a real product? than engineering. Build it and they will come almost never works. Anyway, I think you need a new name for this project. -Original Message- From: Michael Collins m...@freeswitch.org To: freeswitch-users@lists.freeswitch.org freeswitch-users@lists.freeswitch.org; freeswitch-...@lists.freeswitch.org Sent: Tue, 31 Mar 2009 5:10 pm Subject: [Freeswitch-users] Call For Help: Janitor Projects Dear FreeSWITCH Community: As you know, FreeSWITCH has been growing leaps and bounds and it's going to keep growing as the word spreads. The core development team of Anthony, Mike, and Brian are very appreciative of the community's help and involvement in the project. Simply put: the community is awesome! Some have asked how they can help. Most of us are not software developers, but that doesn't mean we can't help to grow the FreeSWITCH ecosystem. To this end I've started a janitor projects wiki page: http://wiki.freeswitch.org/wiki/Janitor_Projects We say janitor projects because they are things that help keep the project clean and organized, just like the janitor cleans an office, takes out the trash, replaces the toilet paper, etc. These are valuable services that we sometimes take for granted. However, I think we can all appreciate that the FreeSWITCH project would be better served if the developers could focus on writing code, fixing bugs, etc. and not on the easier, not-quite-as-important janitorial tasks. To that end we are inviting all who wish to volunteer to please visit the above wiki page and check out some of the projects listed so far. Email me off list if you'd like to volunteer to help. I'm maintaining a list of janitors and what they are helping with. If you have ideas for other janitor projects then by all means email them to me and we'll discuss them. Thanks again for being such a great community! -Michael S Collins IRC: mercutioviz See you at ClueCon 2009!? http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Call For Help: Janitor Projects
I just did, and it was suggestion. -Original Message- From: Larry Edelstein r...@acm.org To: freeswitch-users@lists.freeswitch.org Sent: Wed, 1 Apr 2009 12:00 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects You are then volunteering for something? 2009/3/31 mszla...@aol.com First off. I would not call it a janitors project since that may offend some. A second problem is your notion that documentation is not-quite-as-important a task as writing code. I'm think many would say you have that backwards. There is nothing more effective in evolving FreeSwitch than good documentation which helps further development and is an important part of customer service. Good customer service is then a part of sales and marketing. Much more often than not, It's sales and marketing that is more important to making something a real product? than engineering. Build it and they will come almost never works. Anyway, I think you need a new name for this project. -Original Message- From: Michael Collins m...@freeswitch.org To: freeswitch-users@lists.freeswitch.org freeswitch-users@lists.freeswitch.org; freeswitch-...@lists.freeswitch.org Sent: Tue, 31 Mar 2009 5:10 pm Subject: [Freeswitch-users] Call For Help: Janitor Projects Dear FreeSWITCH Community: As you know, FreeSWITCH has been growing leaps and bounds and it's going to keep growing as the word spreads. The core development team of Anthony, Mike, and Brian are very appreciative of the community's help and involvement in the project. Simply put: the community is awesome! Some have asked how they can help. Most of us are not software developers, but that doesn't mean we can't help to grow the FreeSWITCH ecosystem. To this end I've started a janitor projects wiki page: http://wiki.freeswitch.org/wiki/Janitor_Projects We say janitor projects because they are things that help keep the project clean and organized, just like the janitor cleans an office, takes out the trash, replaces the toilet paper, etc. These are valuable services that we sometimes take for granted. However, I think we can all appreciate that the FreeSWITCH project would be better served if the developers could focus on writing code, fixing bugs, etc. and not on the easier, not-quite-as-important janitorial tasks. To that end we are inviting all who wish to volunteer to please visit the above wiki page and check out some of the projects listed so far. Email me off list if you'd like to volunteer to help. I'm maintaining a list of janitors and what they are helping with. If you have ideas for other janitor projects then by all means email them to me and we'll discuss them. Thanks again for being such a great community! -Michael S Collins IRC: mercutioviz See you at ClueCon 2009!? http://www.cluecon.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org New Low Prices on Dell Laptops - Starting at $399 ___ 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] Call For Help: Janitor Projects
Pardon me, but you speak only for yourself. I think Janitor is not an appropriate word. Second, 'marketing and sales' does not only mean making money. It also means 'selling' someone on the idea of trying something and effectively spreading the word. Third, the original developers can spend most of their time developing because they're the creators so they know very well what's going on with the code and don't need good documentation. Others need good documentation to effectively work with FS or do development. Currently the documentation is scattered, assumes to much and is outdated/incorrected. Also, there is a problem with not getting the creators involved with documentation since someone doing the documentation will have to ask them what's what. The creators never will be totally out of the loop nor should they be. This doesn't apply only here in this context but other similar ones as well. Keeping? creators from inteact with customers is one big reason so many start-ups fail. -Original Message- From: Raul Fragoso r...@etellicom.com To: freeswitch-users@lists.freeswitch.org Sent: Wed, 1 Apr 2009 1:29 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects Pardon my honesty, but I think you are the one who is getting this backwards. Firstly, I fail to see why a call for help with organizing and cleaning up the project documentation would offend someone by simply having janitor as the name. Have you ever heard the term gatekeeper before ? Would it offend you ? Think again. Secondly, FreeSWITCH is an open-source project, so forget the 'marketing sales' crap in the context of documentation. The success of the project, which is growing incredibly fast, is built upon the collaboration of the community as a whole, and it's common sense that sharing the project tasks is a major necessary step to keep it going, just like a janitor is of primordial importance to keep an office building organized and clean. Last but not the least, I agree entirely with the fact that the core developers should be doing what they do it best, and that is, of course, development. I see this call for help request as an effective way of keeping them developing new features and improving the current functionality of FreeSWITCH while sharing the burden of documentation and organization. That's fair and sounds very logical to me. If you join the FreeSWITCH IRC channel and hang in there for a bit you will understand what I mean, most of the time these guys are busy responding to user questions or analyzing use cases that could be easily solved by checking a more organized documentation, and this is what Michael's request is all about. Regards, Raul On Wed, 2009-04-01 at 02:21 -0400, mszla...@aol.com wrote: First off. I would not call it a janitors project since that may offend some. A second problem is your notion that documentation is not-quite-as-important a task as writing code. I'm think many would say you have that backwards. There is nothing more effective in evolving FreeSwitch than good documentation which helps further development and is an important part of customer service. Good customer service is then a part of sales and marketing. Much more often than not, It's sales and marketing that is more important to making something a real product than engineering. Build it and they will come almost never works. Anyway, I think you need a new name for this project. -Original Message- From: Michael Collins m...@freeswitch.org To: freeswitch-users@lists.freeswitch.org freeswitch-users@lists.freeswitch.org; freeswitch-...@lists.freeswitch.org Sent: Tue, 31 Mar 2009 5:10 pm Subject: [Freeswitch-users] Call For Help: Janitor Projects Dear FreeSWITCH Community: As you know, FreeSWITCH has been growing leaps and bounds and it's going to keep growing as the word spreads. The core development team of Anthony, Mike, and Brian are very appreciative of the community's help and involvement in the project. Simply put: the community is awesome! Some have asked how they can help. Most of us are not software developers, but that doesn't mean we can't help to grow the FreeSWITCH ecosystem. To this end I've started a janitor projects wiki page: http://wiki.freeswitch.org/wiki/Janitor_Projects We say janitor projects because they are things that help keep the project clean and organized, just like the janitor cleans an office, takes out the trash, replaces the toilet paper, etc. These are valuable services that we sometimes take for granted. However, I think we can all appreciate that the FreeSWITCH project would be better served if the developers could focus on writing code, fixing bugs, etc. and not on the easier, not-quite-as-important janitorial tasks. To that end we are inviting all who wish to volunteer to please visit the above wiki page and check out some of the projects listed so far. Email me off list if you'd like
Re: [Freeswitch-users] Call For Help: Janitor Projects
Call it what it is like The Documentation Project or something similar. Sure, if there was no code there is no FS but I didn't say the code is not important. I was taking a sales/marketing versus engineering analogy to this and only said that many would find it less important than good documentation if you are looking to get people to use FS and/or evolve the code. So as long as the creators of FS are willing to work to some extent on the documentation with a documentor, when one is needed, then this should work out. The creators have a very good understanding of FS which the documentor may not. On the other hand, the documentor doesn't have the creators background baggage which makes things seem obvious to the creator but isn't to users or even other developers. The creators and documentors working together will hopefully make the FS documentation accurate, not to presumptuous and easy to use. -Original Message- From: Brian West br...@freeswitch.org To: freeswitch-users@lists.freeswitch.org Sent: Wed, 1 Apr 2009 1:42 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects What do you recommend calling it then? ?I wouldn't be offended by it ... and I can't think of any reason it would offend someone because it describes the task at hand. ?As far as documentation vs code... without the code there would be ZERO need for any documentation. ?The code is the hardest part to make sure it functions bug free. ?Developers are great at writing code but not the best at writing documentation, me included. ?It's the perfect place for anyone that wants to help out! ?I welcome anyone and everyone to the project in hopes that community members will help out! ? We have various IRC channels... #freeswitch, #freeswitch-dev, #freeswitch-docs and #freeswitch-social so join irc.freenode.net and get involved because you never know how it might change your life for the better! ;) /b Positive anything is better than negative thinking. On Apr 1, 2009, at 1:21 AM, mszla...@aol.com wrote: First off. I would not call it a janitors project since that may offend some. A second problem is your notion that documentation is not-quite-as-important a task as writing code. I'm think many would say you have that backwards. There is nothing more effective in evolving FreeSwitch than good documentation which helps further development and is an important part of customer service. Good customer service is then a part of sales and marketing. Much more often than not, It's sales and marketing that is more important to making something a real product? than engineering. Build it and they will come almost never works. Anyway, I think you need a new name for this project. Brian West br...@freeswitch.org -- Meet us a ClueCon! ?http://www.cluecon.com = ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Call For Help: Janitor Projects
The holy grail magic documentation that is like the hitchhikers guide to the galaxy or harry potter's marauder's map can tune into what you need to know or what you don't understand and magically adjusts.! Maybe your projecting or exaggerating but I didn't say anything like that. However, the important point was we have a lot of users like that. Enough said. -Original Message- From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Wed, 1 Apr 2009 6:19 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects have a look. http://www.google.com/search?q=janitor+project The phrase has already been coined. If you look closely we have 2 different perspectives in this thread. mszlazak is seeking more of the higher level user documentation, the holy grail magic documentation that is like the hitchhikers guide to the galaxy or harry potter's marauder's map can tune into what you need to know or what you don't understand and magically adjusts.? This is normal,?.? The majority of users will treat us like they are buying the software from us and impose their expectations on us.? It's helpful to us, it lets us see things from their perspective. Seven is looking it at more from a developer's perspective, he's actually willing to take the time to add things to the wiki and he wants to understand how the code works.? This is a good thing too, there are far less people of this type in our community but they are crucial.? Core developers document by explaining what they are doing to people like Seven or by putting a reminder in the commit notes which are later translated into the CHANGELOG for the releases.? Michael, the author of this thread has added countless pages of documentation to the wiki this way.? It's easy to say the author should document everything.? There is close to 300,000 lines of code in just the src directory in the FreeSWITCH tree (that is all code we wrote not counting any of the depends libs or any other form of pre-existing code).? I personally wrote the majority of that code so, I really appricate it when the communiuty gives me a few minutes to take a break while they document it.? The best people to document the high level fuctionality? is not the author btw.? It's the first few people who use it.? Most likely they are developing a product from it and they intend to profit from it in one way or another and its a fair tradeoff to have the section of functionality explained to them in exchange from wikifying it from their perspective.? The perspective of the author will be dry and mechinacal where that first-time-user version of the documentation will make much more sense to future readers.? When it comes to the low level documentation, the C functions, we also need someone to help us with that if they feel there is not enough.? We write code, we know how it works.? If other people cannot figure out how it works, they will ask us and in the end it will be doucmented.? About 5% or less of people in the community even have to look in the code for the core.? The whole point of the FreeSWITCH design is to push everything up to scripts, remote connections and dialplan logic to let people concentrate on good ideas instead of the evil logic necessary to properly engineer a telephony engine.? So I recommend anybody interested starts out making sure there is ample documentation for the embedded and external API for lua, js, perl, python, ESL etc.? Then anybody who really likes C code can start with the module API layer and then dig deeper into the core code and learn how it works and if the documentation is not enough, add some, we appriciate any help we can get. ? 2009/4/1 mszla...@aol.com First off. I would not call it a janitors project since that may offend some. A second problem is your notion that documentation is not-quite-as-important a task as writing code. I'm think many would say you have that backwards. There is nothing more effective in evolving FreeSwitch than good documentation which helps further development and is an important part of customer service. Good customer service is then a part of sales and marketing. Much more often than not, It's sales and marketing that is more important to making something a real product? than engineering. Build it and they will come almost never works. Anyway, I think you need a new name for this project. -Original Message- From: Michael Collins m...@freeswitch.org To: freeswitch-users@lists.freeswitch.org freeswitch-users@lists.freeswitch.org; freeswitch-...@lists.freeswitch.org Sent: Tue, 31 Mar 2009 5:10 pm Subject: [Freeswitch-users] Call For Help: Janitor Projects Dear FreeSWITCH Community: As you know, FreeSWITCH has been growing leaps and bounds and it's going to keep growing as the word spreads. The core development team of Anthony, Mike
Re: [Freeswitch-users] Call For Help: Janitor Projects
nope -Original Message- From: Brian West br...@freeswitch.org To: freeswitch-users@lists.freeswitch.org Sent: Wed, 1 Apr 2009 10:39 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects Are you referring to PocketSphinx here?? /b On Apr 1, 2009, at 12:24 PM, mszla...@aol.com wrote: ?Currently the documentation is scattered, assumes to much and is outdated/incorrected. Brian West br...@freeswitch.org -- Meet us a ClueCon! ?http://www.cluecon.com = ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Call For Help: Janitor Projects
Excellent! The core developers/creators should stay active in the documentation process. -Original Message- From: Brian West br...@freeswitch.org To: freeswitch-users@lists.freeswitch.org Sent: Wed, 1 Apr 2009 10:52 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects On Apr 1, 2009, at 12:45 PM, mszla...@aol.com wrote: Call it what it is like The Documentation Project or something similar. Because its MORE than Documentation! ?So that name is silly! Sure, if there was no code there is no FS but I didn't say the code is not important.?I was taking a sales/marketing versus engineering analogy to this and?only said that many would find it less important than good documentation if you are looking to get people to use FS and/or evolve the code. So as long as the creators of FS are willing to work to some extent on the documentation with a documentor, when one is needed, then this should work out. The creators have a very good understanding of FS which the documentor may not. On the other hand, the documentor doesn't have the creators background baggage which makes things seem obvious to the creator but isn't to users or even other developers. The creators and documentors working together will hopefully make the FS documentation accurate, not to presumptuous and easy to use. Well if people join IRC... ask questions we do answer them... so if people don't understand something all they have to do is ask we won't bite. Brian West br...@freeswitch.org -- Meet us a ClueCon! ?http://www.cluecon.com = ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Call For Help: Janitor Projects
You missed the point again. But suffer fools to long. -Original Message- From: Raymond Chandler intralan...@freeswitch.org To: freeswitch-users@lists.freeswitch.org Sent: Wed, 1 Apr 2009 11:29 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects mszla...@aol.com wrote: Pardon me, but you speak only for yourself. I think Janitor is not an appropriate word. you're welcome to your opinions, no matter how wrong they are Second, 'marketing and sales' does not only mean making money. It also means 'selling' someone on the idea of trying something and effectively spreading the word. we don't try to sell anyone on the project... we'll tell you the pros and cons, you decide if the software meets your needs or not. Third, the original developers can spend most of their time developing because they're the creators so they know very well what's going on with the code and don't need good documentation. Others need good documentation to effectively work with FS or do development. Currently the documentation is scattered, assumes to much and is outdated/incorrected. maybe you could fix some of that since you seem to be very enlightened to its shortcomings? although, that might offend your delicate psyche since you'd basically be a janitor then. Also, there is a problem with not getting the creators involved with documentation since someone doing the documentation will have to ask them what's what. The creators never will be totally out of the loop nor should they be. This doesn't apply only here in this context but other similar ones as well. Keeping? creators from inteact with customers is one big reason so many start-ups fail. hmmm, maybe you're right... maybe the whole idea of hierarchy is entirely wrong. i guess we could expect tony to document his own code... while we're at it, let's suggest that microsoft has Bill Gates write documentation for windows and answer tech support calls, right? cus i mean, obviously everyone who writes code should obviously do everything else too, right? but i guess that doesn't work the other direction... cus if you don't know how to code, then you just can't code... its as simple as that. so now we have effectively halved (or better) the development activities of FreeSWITCH so there's less to document, but that's ok, because now there's plenty of people using it and not contributing anything back... and that's what open-source is really all about, right? btw, i'm just curious if you're an employee of a commercial entity that feels threatened by FreeSWITCH... what better way to decrease productivity than to split hairs over something so stupid as the name of an effort (janitor projects, in this case) that you're not going to take part in anyway. if i may ask, have you done anything constructive for the community at all? all i've seen of you from the mailing lists is non-constructive criticisms. not that we don't appreciate your trolling... its very entertaining to see how narrow-minded some people are. -Ray ___ 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] Call For Help: Janitor Projects
You tried to be nice! Give me a break. Maybe try harder next time. -Original Message- From: Anthony Minessale anthony.miness...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Wed, 1 Apr 2009 11:23 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects Did you follow the link I posted? http://www.google.com/search?q=janitor+project The linux kernel calls it the same thing and so do all the other project that come up in that search. Would you prefer Custodial Engineering projects I tried to be nice but you continue to perpetuate this thread. Another term you may not be familiar with is when someone who is outnumbered starts trying to get the last word on a mailing list or forum, they're called trolls Exactly how much have you contributed to this project other than complaints? You initially contacted us at our consulting address, where we then called you on the phone and helped you for 2 hours for free even though we know your goal is to develop a product from FreeSWITCH and most people in your position offer to pay us for our time.? (make as many products as you want, that's why we made FreeSWITCH so good for you, but, usually if you want *that much* help you have to pay for it) You started using modules that were just written at the time you came around on a platform on which the module only was compiling for a week, give us a break. We have all helped you on the list and documented things *for you* on several dozen occasions. I don't want anything in return but for you to please stop commenting on this thread. This is not a mob rule project, I will make the decisions for it when I see fit and when I seek the input of others, I ask for it and when I don't want any input I do whatever I want.? It's a perk of running your own project.? I personally don't care what Collins calls it, janitor project or whatever, at least he is show initiative and getting people involved. 2009/4/1 mszla...@aol.com Pardon me, but you speak only for yourself. I think Janitor is not an appropriate word. Second, 'marketing and sales' does not only mean making money. It also means 'selling' someone on the idea of trying something and effectively spreading the word. Third, the original developers can spend most of their time developing because they're the creators so they know very well what's going on with the code and don't need good documentation. Others need good documentation to effectively work with FS or do development. Currently the documentation is scattered, assumes to much and is outdated/incorrected. Also, there is a problem with not getting the creators involved with documentation since someone doing the documentation will have to ask them what's what. The creators never will be totally out of the loop nor should they be. This doesn't apply only here in this context but other similar ones as well. Keeping? creators from inteact with customers is one big reason so many start-ups fail. -Original Message- From: Raul Fragoso r...@etellicom.com To: freeswitch-users@lists.freeswitch.org Sent: Wed, 1 Apr 2009 1:29 am Subject: Re: [Freeswitch-users] Call For Help: Janitor Projects Pardon my honesty, but I think you are the one who is getting this backwards. Firstly, I fail to see why a call for help with organizing and cleaning up the project documentation would offend someone by simply having janitor as the name. Have you ever heard the term gatekeeper before ? Would it offend you ? Think again. Secondly, FreeSWITCH is an open-source project, so forget the 'marketing sales' crap in the context of documentation. The success of the project, which is growing incredibly fast, is built upon the collaboration of the community as a whole, and it's common sense that sharing the project tasks is a major necessary step to keep it going, just like a janitor is of primordial importance to keep an office building organized and clean. Last but not the least, I agree entirely with the fact that the core developers should be doing what they do it best, and that is, of course, development. I see this call for help request as an effective way of keeping them developing new features and improving the current functionality of FreeSWITCH while sharing the burden of documentation and organization. That's fair and sounds very logical to me. If you join the FreeSWITCH IRC channel and hang in there for a bit you will understand what I mean, most of the time these guys are busy responding to user questions or analyzing use cases that could be easily solved by checking a more organized documentation, and this is what Michael's request is all about. Regards, Raul On Wed, 2009-04-01 at 02:21 -0400, mszla...@aol.com wrote: First off. I would not call it a janitors project since that may offend some. A second problem is your notion that documentation is
[Freeswitch-users] Help scripting tone_detect.
Except for fields in a dial plan extension, I can't get tone_detect (or stop_tone_detect), http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_stop_tone_detect to work in JavaScript. If there is one, what's the correct syntax for scripting these? 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
[Freeswitch-users] switch_ivr_originate() Parse Error!
I'm getting a parsing error which seems to be coming from the space in
Extension 1000. If this is normal then what's the best way to deal with
spaces in caller id names?
This is how the call was originated:
action application=set data=api_hangup_hook=sched_api +5 none originate
{id_name=${effective_caller_id_name},id_number=${caller_id_number}}sofia/internal/1000%$${domain}
GINO_ANS/??? ??? ???
2009-03-28 10:50:56 [ERR] switch_ivr_originate.c:976 switch_ivr_originate()
Parse Error!
2009-03-28 10:50:56 [DEBUG] switch_ivr_originate.c:2081 switch_ivr_originate()
Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
2009-03-28 10:50:56 [DEBUG] mod_commands.c:2213 sch_api_callback() Command
originate({id_name=Extension 1000,id_number=1000}sofia/internal/1000%10.0.0.3
GINO_ANS):
-ERR DESTINATION_OUT_OF_ORDER
Mark
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Re: [Freeswitch-users] Question about FAQ question
I'd like to do something like Make current but for Windows because I'm finding bugs to report.? One was on how the dialplan's extensions are being parsed. Extensions in some cases like when doing originates with sched_api, loose their last character and I have to add a white space after to solve the problem. Another issue is that dialing one extension landed me into another totally different extension. I had to comment it out to target the right extension. Maybe the reporting bugs wiki needs updating for Windows users (experienced and inexperienced). Thanks. Mark. ? -Original Message- From: Brian West br...@freeswitch.org To: freeswitch-users@lists.freeswitch.org Sent: Wed, 25 Mar 2009 12:07 pm Subject: Re: [Freeswitch-users] Question about FAQ question On Mar 25, 2009, at 2:01 PM, Josh Forman wrote: I'm running Freeswitch on Ubuntu 64bit Intrepid and on a svn rev 12722, freeswitch would install and run fine but as soon as calls started coming in it would have a segmentation fault. This is the second svn snapshot I've had this happen on. Please review http://wiki.freeswitch.org/wiki/Reporting_Bugs Collect the info and report it on jira. If it still happens on SVN trunk as of NOW then please collect the back trace as per the reporting bugs guide and we'll investigate the issue. Also be aware if you aren't doing a make current, you could have build skew which could be the whole problem all along. Incorrectly updating your system will result in all kinds of strange behaviors. In the Freeswitch FAQs there is a question concerning segmentation fault on ubuntu 64bit except they say it occurs on start. Their solution is to recompile libedit which I plan to try regardless, but I just wanted to know if the scenario they refer to is the same as what I'm experiencing or was there some other problem where freeswitch would segfault while initially loading. I don't think this is the problem you're having. Until then I'm still running rev 12289 on the system that is having issues. This is a rather old REV, How about you report your back trace to jira as per the instructions above. Thanks -Josh ___ 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] Question about FAQ question
Thanks MC, maybe a link to that TortoiseSVN would help for some in the Windows crowd. TortoiseSVN has a bunch of stuff in it but to make it simple, especially for doing updates to report bugs, then mentioning if doing just an SVN update will work before rebuild. Also, what to do if one gets an errror using SVN Update. I got one about not being able to open a file. So, I didn't know if any of the rest of the SVN update succeeded. I guess it wasn't a clean update. I didn't bother rebuilding afterwards since I also didn't know if it would work. There was to much to go through in TortoiseSVN documentation for the time I had so I didn't report the errors and left things for later when I would just download and install a newer version of FS. It sounds lazy but as an inexperienced user it's enough discouragment to let these things go. Anyway, just a bit more instructions might get more bugs reported. Thanks again. Mark. -Original Message- From: Michael Collins m...@freeswitch.org To: freeswitch-users@lists.freeswitch.org Sent: Wed, 25 Mar 2009 12:59 pm Subject: Re: [Freeswitch-users] Question about FAQ question Maybe the reporting bugs wiki needs updating for Windows users (experienced and inexperienced). Quite possibly. The instructions are not explicit. I will add something for the Windows users that's a bit more specific. -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ 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] setInputCallback not working with Javascript?
I'm getting in build 12653M: [ERR] notify.js:130 mod_spidermonkey()? TypeError: session.setInputCallback is not a function The wiki says this function should work in Javascript. http://wiki.freeswitch.org/wiki/CoreSession_Constructor#session:setInputCallback Also, has there been changes to session.collectInput with type=event? I get dtmf type events with my callback function but can't seem to get type=event with speech events. 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
Re: [Freeswitch-users] Cepstral and RSS feeds
This product has much better sounding TTS than Cepstral: http://www.neospeech.com/Default.aspx Maybe you can use Dave's suggestion and make WAV recordings from their demo text input box and then just clip off the initial portion. Mark. -Original Message- From: David Knell d...@3c.co.uk To: freeswitch-users@lists.freeswitch.org Sent: Fri, 20 Mar 2009 6:53 pm Subject: Re: [Freeswitch-users] Cepstral and RSS feeds In the meantime, you can work around this by using the swift executable to turn text in to WAV files, and then just play them back.? Works fine for short(ish) texts - there might be a bit of a delay if you wanted the thing to read back War and Peace. --Dave http://jira.freeswitch.org/browse/MODASRTTS-11 Might wanna know about that issue also :) /b On Mar 20, 2009, at 12:02 AM, Carlos Talbot wrote: I wrote this wiki page a while back. Did it help? http://wiki.freeswitch.org/wiki/Mod_rss ___ 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] Is mod vmd working?
I put this after the vmd tag action application=playback data=SIT/IO_SIT.wav/ to check vmd with tones found on this page http://en.wikipedia.org/wiki/Special_information_tone I converted them over with Audacity to wav files and vmd worked in finding a beep but the format was wrong for FS. However, after I switch the format of the audio files to something FS likes then vmd would not detect the tones. Is there some good test tones for the U.S phone system I could use to check both mod_vmd and tone_detect? Thanks. Mark. -Original Message-From: Michael Collins m...@freeswitch.org To: freeswitch-users@lists.freeswitch.org Sent: Thu, 19 Mar 2009 8:48 am Subject: Re: [Freeswitch-users] Is mod vmd working? tone_detect! sounds good. BTW, was there any errors in those extensions I posted. I modified something you posted MC. Not at first glance. What did you change? -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ 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] Feedback on Freeswitch for Windows?
I'm doing what you want to do and using SPA3102. It's much easier to get someone to try it this way when dealing with small mom and pop size business. Haven't tried higher concurrent call volumes with some of the PCI cards mentioned. If you haven't done this already, my advice is first to see whether there is enough of a market for you and what the issues will be with potential customers before investing to much time on the technical side. That means putting on your suit on and visiting businesses for a few months with a notebook. I'll tell you some things, first their maybe many businesses that don't want an internet connection and don't even bother mentioning voip if you want to resell that service. The rest you'll find out. Good luck. Mark. -Original Message- From: Gilles codecompl...@free.fr To: freeswitch-users@lists.freeswitch.org Sent: Thu, 19 Mar 2009 3:45 pm Subject: [Freeswitch-users] Feedback on Freeswitch for Windows? (sorry for the broken thread: I don't know how to avoid this when answering through the digest version of the mailing list) Michael Jerris You could use Netborder Express with it. Thanks for the tip. I didn't know this device. I'm not sure I understand the difference between this PCI card and other Sangoma PCI cards that offer an FXO port, though :-/ mercutioviz Is there a compelling reason to use a Windows machine? Yes. I'd like to offer a really cheap solution for those customers who don' t mind using their workstation as Freeswitch IVR server, so I can just provide a Linksys VoIP gateway and the software for Windows, and they're ready to go. I'll go ahead and play with the Windows port of Freeswitch, and see how it goes. Thank you. ___ 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] Is mod vmd working?
I followed these instructions for Mod_vmd except for a Windows box: http://wiki.freeswitch.org/wiki/Mod_vmd I tried testing to see if it's working by dialing the following extension: ??? !-- mod_vmd test extension (new mod)-- ??? extension name=vmdtest ??? ??? condition field=destination_number expression=^$ ??? ??? ??? action application=answer/ ??? ??? ??? action application=info/ ??? ??? ??? action application=vmd/ ??? ??? ??? action application=sleep data=25000/ ??? ??? ??? action application=info/ !-- Look for chan var vmd_detect here -- ??? ??? ??? action application=hangup/ ??? ??? /condition ??? /extension However, I didn't see channel variable vmd_detect in the FreeSwitch console. ?? 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
Re: [Freeswitch-users] Is mod vmd working?
SDR?
I'm wondering why there was nothing in the console showing the channel
variable ${vmd_detect} as the wiki says there should be:
action application=info/ !-- Look for chan var vmd_detect here --
Mark
-Original Message-
From: Shelby Ramsey sicfsl...@gmail.com
To: freeswitch-users@lists.freeswitch.org
Sent: Wed, 18 Mar 2009 12:11 am
Subject: Re: [Freeswitch-users] Is mod vmd working?
Mark,
It does work ... but I can't really attest to how well ... especially compared
to other things out there.? I started capturing this in CDR's to see and it
didn't seem like it worked very well.
If this is really critical to you, you might want to ping Ken Rice.? I know he
might have a better option.
SDR
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Re: [Freeswitch-users] Is mod vmd working?
I added a voicemail tag in to a default extension 1001, I hear the voicemail beep but still don't see vmd_detect. Mark -Original Message- From: Shelby Ramsey sicfsl...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Wed, 18 Mar 2009 6:07 am Subject: Re: [Freeswitch-users] Is mod vmd working? Mark, Because it didn't detect a beep.? It will be be there as vmd_detect=true if it does.? I'm not sure exactly how reliable it's beep detection is.? SDR ___ 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] Is mod vmd working?
Hi MC,
With trunk 12638M, I tried checking vmd internally and externally to my cell.
No luck at all in detecting a voicemail (beep).
I used the following extensions to test this, maybe they are in error.
If not then how else can I detect from FS that I got voicemail in a phone
agnostic way (i.e, pots sip).
extension name=110
condition field=destination_number expression=^110$
action application=answer/
action application=vmd/
!--action application=voicemail data=default 10.0.0.3 1000/--
action application=bridge
data=sofia/gateway/spa3102PSTN/12223334...@10.0.0.5:5061/
action application=transfer data=111 XML default/
/condition
/extension
extension name=111
condition field=destination_number expression=^111$/
condition field=${vmd_detect} expression=^TRUE
action application=answer/
action application=speak data=flite|kal|voicemail detected/
action application=hangup/
anti-action application=answer/
anti-action application=speak data=flite|kal|no voicemail detected/
anti-action application=hangup/??? ??? ???
/condition
/extension
Mark.
-Original Message-
From: Michael Collins m...@freeswitch.org
To: freeswitch-users@lists.freeswitch.org
Sent: Wed, 18 Mar 2009 10:24 am
Subject: Re: [Freeswitch-users] Is mod vmd working?
2009/3/18 mszla...@aol.com:
I added a voicemail tag in to a default extension 1001, I hear the
voicemail beep but still don't see vmd_detect.
Mark
FYI, I've used mod_vmd but only in a TDM environment on outbound calls
via a PRI. It worked very well on for detecting answering machine
beeps and vm beeps on cell phone voice mails.
-MC
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Re: [Freeswitch-users] Is mod vmd working?
tone_detect! sounds good. BTW, was there any errors in those extensions I posted. I modified something you posted MC. -Original Message- From: Michael Collins m...@freeswitch.org To: freeswitch-users@lists.freeswitch.org Sent: Wed, 18 Mar 2009 5:15 pm Subject: Re: [Freeswitch-users] Is mod vmd working? Ironically, I've used tone_detect to try and trap SIT tones and I found that answering machines in the USA seem to all send a beep in the same freq range as American SIT tones... :) -MC On Wed, Mar 18, 2009 at 4:22 PM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hmm, Well We're connected direct to E1's and it doesn't work reliably here. That said, DTMF detect does recognise the beeps most of the time. Perhaps there's a regional variation. ?I wonder if it's country specific. ?The code looks logical. ?When I get some time I'll have a look at it and see how it can be improved. The concept is great and is much better that sniffing out human voice as that's prone to false positives. ?Much better to assume human and machine. ?Nothing worse than a silent call. Regards, -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 18 March 2009 17:24 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Is mod vmd working? 2009/3/18 ?mszla...@aol.com: I added a voicemail tag in to a default extension 1001, I hear the voicemail beep but still don't see vmd_detect. Mark FYI, I've used mod_vmd but only in a TDM environment on outbound calls via a PRI. It worked very well on for detecting answering ?machine beeps and vm beeps on cell phone voice mails. -MC ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ 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] Problem with detecting session.cause with Linksys adaptor
Hello,
I have an analogue line connected to SPA3102 which then sends calls to
my FreeSwitch (FS) application. I can originate a call from FS and dial
out the analogue line almost fine. However, an issue seems to be
detecting if the PSTN line is busy. Another issue might be hanging up
the PSTN side of the adaptor from FS side.
Anyway here's the problem. If I then try originating an outbound call from FS
through the pstn
side of the adaptor and if a caller on the analogue line has not
hung up then they hear FS dialing and any automated message that FS then
sends. I can not seem to detect from FS if the PSTN line is busy for
some reason with the following code.
var s;
while (tryCalling()) {}
s.hangup();
exit();
function tryCalling() {
??? s = new Session(sofia/gateway/spa3102/14082031...@10.0.0.5:5061);
??? s.waitForAnswer(3);
???
??? if (s.cause == USER_BUSY) {
??? ??? return true;
??? }
??? if (s.ready()) {
??? ??? s.sleep(1000);
??? ??? s.speak(cepstral,Callie,Hello from Gino Mick Gelato);
??? }
??? return false;
}
I've also tried the above with ignore_early_media=true but no luck.
Thanks.
Mark.
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[Freeswitch-users] How do I notify FreeSwitch that a phone has been answered to play audio or TTS
If I originate an outgoing call from FreeSwitch and want to tts a phrase or
play an audio once the call has been answered (i.e, someone answered their cell
phone or I got their voicemail) then how do I detect that.
Otherwise, I've tried the following but it relies in getting the timing right
which won't always work or looping the tts phrase over and over.
var s;
while (tryCalling()) {}
s.hangup();
exit();
function tryCalling() {
??? s = new Session(sofia/gateway/spa3102/12223334...@10.0.0.5:5061);
??? s.waitForAnswer(1);
???
??? if (s.cause == USER_BUSY) {
??? ?? ?return true;
??? }
??? if (s.ready()) {
??? ?? ?s.sleep(1);
??? ?? ?s.speak(cepstral,Callie,Hello from FreeSwitch);
??? }
??? return false;
}
Another way is to keep replaying the tts phrase by replacing
? if (s.ready()) {
??? ?? ?s.sleep(1);
??? ?? ?s.speak(cepstral,Callie,Hello from FreeSwitch);
??? }
with something like:
while (s.ready()) {
??? ?? ?s.speak(cepstral,Callie,Hello from FreeSwitch);
??? }
Thanks. Mark.
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[Freeswitch-users] Help with detecting hangup events.
I have a problem in setting up an extension that will detect different
hangup events.
If an outside call comes in and gets sent to myExtension then that call is
bridged to an application. When the application hangs up then the hangup
hook notify.js is used to originate a call to another phone which notifies
that phone that the application has been used. However, the problem here is
that in the initial call, the caller can't hangup because then notify.js
won't execute.
extension name=myExtension
condition field=destination_number expression=^$
action application=info/
action application=set data=bypass_media=true/
action application=set
data=effective_caller_id_number=${caller_id_number}/
action application=set data=hangup_after_bridge=true/
action application=set data=api_hangup_hook=jsrun
notify.js/
action application=bridge
data=sofia/internal/${sip_to_us...@127.0.0.1:5068/
/condition
/extension
I haven't tried this but if I changed things so hang_after_bridge=false
then I'm guessing that notify.js will execute when the initial caller hangs
up their phone. This I want but I don't want the call originated in
notify.js to happen unless the bridged to application also issued a hangup
previously. Otherwise, if the caller hangs up before the application does
it's thing and hasn't issued a hangup then notify.js will still originate
calls and these I don't want.
So, I want notify.js to originate calls after the caller hangs up but these
hangups have to be those that came after the bridged to application issued a
hung up.
I hope this makes sense.
Thanks.
Mark.
--
View this message in context:
http://www.nabble.com/Help-with-detecting-hangup-events.-tp22507114p22507114.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.
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Re: [Freeswitch-users] FreeSWITCH forum community opened today!
Great, I like forums better than lists. Make sure the folks at FreeSwitch make it know to anyone coming to any of their pages by providing links, etc. Thanks -Original Message- From: Harry FSwitch switchser...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Thu, 12 Mar 2009 11:55 am Subject: [Freeswitch-users] FreeSWITCH forum community opened today! Greetings, Last week I submitted this post to the mailing list... http://www.nabble.com/Please-end-the-torment-td2235.html I received a mixed response to say the least, no one however emailed me saying they wanted to be involved or would contribute hosting resources. I figured eh screw-it, and went ahead and created the forums anyway and opened them today! http://freeswitch411.info I went on IRC to let folks know its available and after a few jabs from the crowd Anthony said: #freeswitch 2009-03-12 10:47:04 [anthm] harr, if you maintain it you are welcome to have it If it takes off and provides a friendly, helpful entryway for new FreeSWITCH users then I will be happy, if it flops I'll be said. But I will maintain it and do what I can to help FreeSWITCH grow. This is the only time I'll overtly mention it on this list, I'll have the link in my signature and thats about it. :) Thanks for your attention -- Harry http://freeswitch411.info ___ 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] Problem with continue in extension.
I have the following in my dialplan.
Individually, each extension does what it's suppose to do when dialing .
However, if I place continue=true in the first extension then it alone gets
executed and the succeeding extension does not.
I thought condition=true would allow the extension afterward to execute.
My test hardware for this dialplan is a single PSTN line.
A call comes in that line and myExtension executes then hopefully hangs up to
free the line.
Afterward, I want myExtension_Continued to execute the .js application and
dial out that single PSTN line.
I need help in getting this scenerio to work.
Thanks.
?? ?
?? ?extension name=myExtension continue=true?? ?
?? ??? ?condition field=destination_number expression=^$
?? ??? ??? ?action application=info/
?? ??? ??? ?action application=set data=bypass_media=true/
?? ??? ??? ?action application=set
data=effective_caller_id_number=${caller_id_number}/
?? ??? ??? ?action application=set data=hangup_after_bridge=true/
?? ??? ??? ?action application=bridge
data=sofia/internal/${sip_to_us...@127.0.0.1:5068/
?? ??? ?/condition
?? ?/extension
?? ?extension name=myExtension_Continued?? ?
?? ??? ?condition field=destination_number expression=^$
?? ??? ??? ?action application=javascript data=myScript.js/
?? ??? ?/condition
?? ?/extension
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Re: [Freeswitch-users] Problem with continue in extension.
I changed that tag so that hangup_after_bridge is false:
?action application=set data=hangup_after_bridge=false/
but I still don't get the .js application working which is nothing more than a
test script that ran if I dialed it's extension with the preceding one
commented out:
s = new
Session({ignore_early_media=true}sofia/gateway/spa3102/12223334...@10.0.0.5:5061);
while (s.ready()) {
??? s.answer();
??? s.speak(cepstral,Callie,Hello World);
}
-Original Message-
From: Michael Jerris m...@jerris.com
To: freeswitch-users@lists.freeswitch.org
Sent: Wed, 11 Mar 2009 4:34 am
Subject: Re: [Freeswitch-users] Problem with continue in extension.
The continue works fine, it just hangs up befoer that due to:
?? ? ? ? ? ?action application=set data=hangup_after_bridge=true/
Mike
On Mar 11, 2009, at 2:46 AM, mszla...@aol.com wrote:
I have the following in my dialplan.
Individually, each extension does what it's suppose to do when dialing .
However, if I place continue=true in the first extension then it alone gets
executed and the succeeding extension does not.
I thought condition=true would allow the extension afterward to execute.
My test hardware for this dialplan is a single PSTN line.
A call comes in that line and myExtension executes then hopefully hangs up
to free the line.
Afterward, I want myExtension_Continued to execute the .js application and
dial out that single PSTN line.
I need help in getting this scenerio to work.
Thanks.
?? ?
?? ?extension name=myExtension continue=true?? ?
?? ??? ?condition field=destination_number expression=^$
?? ??? ??? ?action application=info/
?? ??? ??? ?action application=set data=bypass_media=true/
?? ??? ??? ?action application=set
data=effective_caller_id_number=${caller_id_number}/
?? ??? ??? ?action application=set data=hangup_after_bridge=true/
?? ??? ??? ?action application=bridge
data=sofia/internal/${sip_to_us...@127.0.0.1:5068/
?? ??? ?/condition
?? ?/extension
?? ?extension name=myExtension_Continued?? ?
?? ??? ?condition field=destination_number expression=^$
?? ??? ??? ?action application=javascript data=myScript.js/
?? ??? ?/condition
?? ?/extension
=
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Re: [Freeswitch-users] Problem with continue in extension.
Mike, no luck with that either.
I still need to see this through but another related approach will be needed
later so I'll ask now.
Does FreeSwitch have some script or something to set up an auto dialer.
Basically, I want to be able the store some caller info then have FS
automatically check to see if a reminder calls need to be sent out.
Thanks.?
-Original Message-
From: Michael Collins m...@freeswitch.org
To: freeswitch-users@lists.freeswitch.org
Sent: Wed, 11 Mar 2009 10:14 am
Subject: Re: [Freeswitch-users] Problem with continue in extension.
On Wed, Mar 11, 2009 at 9:42 AM, mszla...@aol.com wrote:
I changed that tag so that hangup_after_bridge is false:
?action application=set data=hangup_after_bridge=false/
but I still don't get the .js application working which is nothing more than
a test script that ran if I dialed it's extension with the preceding one
commented out:
Try adding a break=never to your first extension:
extension name=myExtension continue=true
condition field=destination_number expression=^$ break=never
action application=info/
action application=set data=bypass_media=true/
action application=set
data=effective_caller_id_number=${caller_id_number}/
action application=set data=hangup_after_bridge=true/
action application=bridge
data=sofia/internal/${sip_to_us...@127.0.0.1:5068/
/condition
/extension
I believe that will cause the the dialplan to keep looking for
even after it has been matched once.
-MC
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Re: [Freeswitch-users] Please end the torment
If as you say people prefer forums then that's the nature of the target market and controlling markets can be very difficult. So you go with the market to succeed. The build it and they will come attitude virtually never works well. -Original Message- From: Kristian Kielhofner kristian.kielhof...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Thu, 5 Mar 2009 12:39 pm Subject: Re: [Freeswitch-users] Please end the torment A bunch of telephony geeks and a 1900 number - what could go wrong? Anyways, I too don't understand why people prefer forums. I follow dozens of mailling lists and a half a dozen e-mail addresses without ever leaving my mail client. My mail client happens to be gmail, btw: - Much more customization, filtering, etc possible than any web forum - Local copies of all messages - Search is awesome, ever hear of Google? ;) Web forums are good when you have to serve ads to people to get paid. Other than that they are certainly not the ideal tool for the job. Besides (and don't take this as an insult) - have you ever compared the web forums to the mailing lists for projects that offer both? Say what you want to say about mailing lists and IRC but the reality (usually) is the l33tz all hang out here and web forums (almost always) end up with the same groups of n00bz circling around and around trying to figure out how to accomplish even the most basic of tasks. Obviously that can go both ways but as a rule of thumb the people that are usually in a position to help others typically prefer mailing lists (probably for some of the reasons I cited above). Or maybe they are just old gray hairs too stuck in their ways. I don't know. ;) On Thu, Mar 5, 2009 at 12:43 PM, Gregory Boehnlein da...@nacs.net wrote: You guys should setup 1-900-FREESWITCH w/ a $1 / minute charge.. :) ___ 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 -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Please end the torment
Again, if your target market prefers lists, then go with list. If they prefer forums then it's forums. The point is that it's not about what a few like, it's about the mob but the right mob. -Original Message- From: David Dan davidw...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Thu, 5 Mar 2009 5:26 pm Subject: Re: [Freeswitch-users] Please end the torment Web forums are like the wild west of the internet.? They offer nothing that a mailing list and good wiki can't handle. Just go take a look at the trixbox forums. The last thing you want it for someone that is looking into freeswitch for the first time, to come across something like this (The Beginning of the End for CE), 1 click off the front page.? I'd really hate to see FS go down this Mob Rule path. On Thu, Mar 5, 2009 at 8:01 PM, Jason White ja...@jasonjgw.net wrote: mszla...@aol.com mszla...@aol.com wrote: If as you say people prefer forums then that's the nature of the target market and controlling markets can be very difficult. So you go with the market to succeed. Most free software/open-source people I've encountered prefer mailing lists and don't like being forced to use a Web interface instead (unless it's the Web interface of their preferred Web mail provider, in which case they're not being compelled to use it). For some of us, a Web forum is hard and inconvenient to use, because it substitutes the forum operator's user interface for that of the user's preferred mail client. I have reasons for choosing the mail client that I use, and if I had to work via somebody else's Web interface instead it would probably result in my not participating at all. This list can also be accessed via the Web and over NNTP at gmane.org. For NNTP enthusiasts, the news group is mane.comp.telephony.freeswitch.user - just connect your news reader to news.gmane.org. You can also post from the newsgroup; the first time you do so, an automated e-mail message will arrive in your inbox requesting confirmation, for spam prevention purposes. I don't know whether it is possible to post from the gmane.org Web site. They use Xapian as their search tool, which, in my experience, usually places the most relevant posts near the top of the search results. ___ 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] pocketsphinx and event socket
Brian, Peter says: mod_pockesphinx has changed/evolved significantely Since this seems to be coming without any warning, what specifically are all these and future changes and why are they happening? Mark. -Original Message- From: Brian West br...@freeswitch.org To: freeswitch-users@lists.freeswitch.org Sent: Tue, 3 Mar 2009 7:00 pm Subject: Re: [Freeswitch-users] pocketsphinx and event socket Well you should use ESL then ;) /b On Mar 3, 2009, at 7:05 PM, Peter P GMX wrote: Thank you Brian, I will try this later. Currently I was happy to get this working on SVN 10003. As : Will there also be major changes in the events I receive through mod_eventsocket? I spend some time on parsing the right data out of the eventsocket interface, and I would just have an idea, if I will have to expect significant work to do, when I later switch to the current SVN. Will I need updated grammar files for the other models too? Best regards Peter ___ 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] pocketsphinx and event socket
I think you need to talk to Brian. Apparently this is a new pocketsphinx which works on a different format from those found in the pizza demo. Also, pocketsphinx crashes if it hears anything outside the grammar which apparently is a longstanding bug. Brian mentioned they are working on getting this fixed. I kept getting: 2009-02-25 19:49:32 [ERR] mod_pocketsphinx.c:140 pocketsphinx_asr_load_grammar() Can't open dictionary C:\Source\freeswitch-snapshot\Debug\grammar\default.dic. 2009-02-25 19:49:32 [WARNING] mod_pocketsphinx.c:219 pocketsphinx_asr_close() Port Closed. The suggestion was to Just copy the cmudict.0.6d to default.dic, not sure how well it will perform on windows.. if it does badly you can slim the dictionary down to words you know you'll be using. https://cmusphinx.svn.sourceforge.net/svnroot/cmusphinx/trunk/cmudict/cmudict.0.6d That gave me more problems so I'm waiting for the fix. Mark. -Original Message- From: Peter P GMX prometheus...@gmx.net To: freeswitch-users@lists.freeswitch.org Sent: Mon, 2 Mar 2009 3:42 pm Subject: Re: [Freeswitch-users] pocketsphinx and event socket Thanks Addison. The Pizza files are there (as mentionned is it a copy of an already working system). In fact freeswitch is complaning about /usr/local/freeswitch/grammar/model/communicator which he cannot load So somehow freeswitch is not willing to open the files, but I have no clue why. So any hints are welcome. Best regards Peter Addison Martin schrieb: Peter, You need the grammar files for the pizza demo: http://wiki.freeswitch.org/wiki/Mod_pocketsphinx#Testing_with_the_Pizza_Demo has lonks to premade fles for everyhting to get the pizza demo working with pocketshinx. Those to not come with the source code when you update from SVN. Nik On Mon, Mar 2, 2009 at 2:31 PM, Peter P GMX prometheus...@gmx.net wrote: Some more info: the system I am working on is a copy (dd copy) of a system where the pizza demo works on. The only thing I changed was to update to the current freeswitch trunk 12293 (it was 10003 before). Do I need to update the model? I did a make in the model directory, but no change. Best regards Peter Peter P GMX schrieb: Hello Brian, thanks for the info. I am a step further, but it cannot load the grammar files. I am sending through event_socket: SendMsg call-command: execute execute-app-name: detect_speech execute-app-arg: pocketsphinx yes no However I get the message (also when I am using Pizza demo): 2009-03-01 23:02:24 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event() sofia/internal/1...@sip2.server.com Command Execute detect_speech(pocketsphinx yes no) 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:145 pocketsphinx_asr_load_grammar() Can't open language model /usr/local/freeswitch/grammar/model/communicator. 2009-03-01 23:02:24 [DEBUG] switch_ivr_async.c:2041 switch_ivr_detect_speech() Error loading Grammar 2009-03-01 23:02:24 [WARNING] mod_pocketsphinx.c:219 pocketsphinx_asr_close() Port Closed. However the grammar files are there: r...@sip2:/usr/local/freeswitch/grammar/model/communicator# r...@sip2:/usr/local/freeswitch/grammar/model/communicator# ls -al total 12752 drwxr-xr-x 2 freeswitch root 4096 2008-08-13 16:00 . drwxr-xr-x 4 freeswitch root 4096 2008-08-13 16:00 .. -rw-r--r-- 1 freeswitch root 1775 2008-03-21 23:32 COPYING -rw-r--r-- 1 freeswitch root 169 2008-03-21 09:21 feat.params -rw-r--r-- 1 freeswitch root 6476668 2008-03-21 09:21 mdef -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 means -rw-r--r-- 1 freeswitch root 263 2008-03-21 15:24 noisedict -rw-r--r-- 1 freeswitch root 6406784 2008-03-21 10:07 sendump -rw-r--r-- 1 freeswitch root 6184 2008-03-21 10:07 transition_matrices -rw-r--r-- 1 freeswitch root 52304 2008-03-21 10:07 variances Any hint? Best regards Peter Brian West schrieb: You can accomplish this here is an example using ESL in perl http://fisheye.freeswitch.org/browse/FreeSWITCH/libs/esl/perl/server3.pl?r=12344 /b On Feb 28, 2009, at 6:18 AM, Peter P GMX wrote: Or back to the basics: Is it possible to use pocketsphinx through event socket? ___ 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] New build gives error message for default grammar file??
I'm getting this error message trying out the pizza demo in FS 1.0.3: ?Can't open dictionary C:\Program Files\FreeSWITCH\grammar\default.dic I didn't have this before where there was no default.dic file. Is there some place a path has to be set now? Thanks. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] New build gives error message for default grammar file??
Hi Brian, It sounds like I'd be better off with 1.0.3 than SVN and will waiting for the fix? But thanks for the files and info. Mark. -Original Message- From: Brian West br...@freeswitch.org To: freeswitch-users@lists.freeswitch.org Sent: Tue, 24 Feb 2009 11:45 am Subject: Re: [Freeswitch-users] New build gives error message for default grammar file?? You're not in 1.0.3 you're in SVN trunk... The reason I know this is that wasn't changed till AFTER 1.0.3 was tagged and a new file set was used... please go into your libs dir and wipe out pocketsphinx and sphinx base.. then let it redownload them. ?I'll make you a new tarball of the new grammar files which are in the jsgf format. ?An example was added to scripts yes_no.gram ... word of warning the new pocketsphinx loads the entire dictionary on port open which on a machine of any speed should load the entire thing in 2 seconds or less... but here is the WARNING WARNING WARNING... If you happen to use words that aren't in the dictionary pocketsphinx WILL CRASH.. I am already on this with the dev's to work out a fix for this its been a long standing bug apparently in the lib. /b On Feb 24, 2009, at 1:36 PM, mszla...@aol.com wrote: I'm getting this error message trying out the pizza demo in FS 1.0.3: ?Can't open dictionary C:\Program Files\FreeSWITCH\grammar\default.dic I didn't have this before where there was no default.dic file. Is there some place a path has to be set now? Thanks. = ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] New build gives error message for default grammar file??
Hey Brian, Where abouts do you keep the Window MSI 1.0.3 build that isn't in SVN trunk. Installing from the wiki installation page gets me a build with the same error. Thanks. Mark. -Original Message- From: Brian West br...@freeswitch.org To: freeswitch-users@lists.freeswitch.org Sent: Tue, 24 Feb 2009 11:45 am Subject: Re: [Freeswitch-users] New build gives error message for default grammar file?? You're not in 1.0.3 you're in SVN trunk... The reason I know this is that wasn't changed till AFTER 1.0.3 was tagged and a new file set was used... please go into your libs dir and wipe out pocketsphinx and sphinx base.. then let it redownload them. ?I'll make you a new tarball of the new grammar files which are in the jsgf format. ?An example was added to scripts yes_no.gram ... word of warning the new pocketsphinx loads the entire dictionary on port open which on a machine of any speed should load the entire thing in 2 seconds or less... but here is the WARNING WARNING WARNING... If you happen to use words that aren't in the dictionary pocketsphinx WILL CRASH.. I am already on this with the dev's to work out a fix for this its been a long standing bug apparently in the lib. /b On Feb 24, 2009, at 1:36 PM, mszla...@aol.com wrote: I'm getting this error message trying out the pizza demo in FS 1.0.3: ?Can't open dictionary C:\Program Files\FreeSWITCH\grammar\default.dic I didn't have this before where there was no default.dic file. Is there some place a path has to be set now? Thanks. = ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Is anyone our there using an SPA 3000? (or 3102)
I'm using and SPA3102 at home to connect one of the phones to my PC.
Nothing special to FS directory files.
I just copied one of the defaults and changed the file name to line1 since I
use line1 as the ID under subscriber information in 3102's configuration.
include
? user id=line1 mailbox=line1
??? params
? param name=password value=1234/
? param name=vm-password value=line1/
??? /params
??? variables
? variable name=toll_allow value=domestic,international,local/
? variable name=accountcode value=line1/
? variable name=user_context value=default/
? variable name=effective_caller_id_name value=Extension line1/
? variable name=effective_caller_id_number value=line1/
? variable name=outbound_caller_id_name
value=$${outbound_caller_name}/
? variable name=outbound_caller_id_number
value=$${outbound_caller_id}/
? variable name=callgroup value=techsupport/
??? /variables
? /user
/include
3102 has a static IP address.
Line 1 has proxy set to FreeSwitch IP.
3102's line 1 tab has dial plan:
(xx.:@gw0)
PSTN line dial plans are for line 1:
?(:2007S0)
and the rest are:
(xx.)
then
Line 1 VoIP Caller DP: 2
VoIP Caller Default DP:1
-Original Message-
From: Scott Ellis scott.el...@novatex.com.au
To: freeswitch-users@lists.freeswitch.org
Sent: Mon, 19 Jan 2009 12:55 pm
Subject: [Freeswitch-users] Is anyone our there using an SPA 3000? (or 3102)
If so, could you please share your set up?
directory files, and dial plan details (gateway details if configured
this way)?
Scott
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Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl
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.miness...@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, mszla...@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 br...@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 ATT 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, mszla...@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!
Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl
Hi Paul, If you mean fixing up pocketsphinx (ps) for telephony instead of or in addition to working on unimrcp then this is the site of the person who created ps and he may have some advice. http://www.cs.cmu.edu/~dhuggins/ Also, this was a post from the sphinx forums for adapting pocketsphinx for telephony. http://sourceforge.net/forum/message.php?msg_id=5621913 I don't know how accurate it is but if accurate then here is that post to give you some of the issues involved: - Well, there are issues in both the decoder and the interface with the telephony application. ? First about the decoder, pocketsphinx right now is the most supported and most feature-reach decoder of the family, but in general it's still oriented on the embedded devices. For telephony applications you probably need to extend it a lot. The features that are currently missing are probably: ? * Out-of-box support for multiple recognizers (probably more a freeswitch issue and a model training issue, for example we have no free male/female model). ? ? * Speaker clustering. ? ? * Automatic VTLN estimation from pitch (This looks simple). ? ? * Good endpointer. ? ? * Discriminative training support in SphinxTrain (Huge task). ? * Good and clean support for a garbage model to be able to filter out out of grammar words. ? * Embedded RASTA extraction and RASTA model training. ? * Advanced features extraction ? Another issue is dialog tracking and understanding. CMU folks are doing work on dialog systems, for example Raven is available ? http://www.ravenclaw-olympus.org/systems_overview.html ? It would be worth to look on it and try to integrate it into freepbx. Decoder will need to support combined language model. As well as you'll need a component for postprocessing. The postprocessing includes disfluency removal, text normalization, text boundary detection. Integration with nltk probably useful for sense extraction. ? If you need more details on any of the above, feel free to ask. --- -Original Message- From: Paul Herring pa...@instruments.com To: freeswitch-users@lists.freeswitch.org Sent: Tue, 13 Jan 2009 8:18 am Subject: [Freeswitch-users] FreeSWITCH, MRCP and Perl What would it take to put a budget together to for this project? Date: Tue, 13 Jan 2009 01:55:36 -0500 From: mszla...@aol.com Subject: Re: [Freeswitch-users] FreeSWITCH, MRCP and Perl To: freeswitch-users@lists.freeswitch.org Message-ID: 8cb436312a08329-80c-1...@mblk-m24.sysops.aol.com Content-Type: text/plain; charset=us-ascii 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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ 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] FreeSWITCH, MRCP and Perl
Great! I hope you will try doing Voxeo's Prophecy next as well ;-) Thanks Dave. Mark. -Original Message- From: David Knell d...@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 ___ 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] FreeSWITCH, MRCP and Perl
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 br...@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 ___ 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] FreeSWITCH, MRCP and Perl
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 br...@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 1 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, mszla...@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 ___ 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] FreeSWITCH, MRCP and Perl
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 br...@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 ATT 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, mszla...@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 ___ 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] Can I transfer information from local application to FS without the need of another server?
I would like to transfer information from an ASR application to FS (both on the same box) without setting up another server (e.g. PHP+Apache). I see that FS can do outgoing HTTP request but does it have something for handling inbound requests? I thought of transferring from the other app a to FS extension, have FS do some Javascript, then transfer back to the app but no information is passed except that it maybe implicit in the extension that is chosen. Thanks. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] How do I set my FS internal ip address to a static value.
I believe there is a variable $${domain} that is used by FreeSwitch for it's
internal IP address. This address was 10.0.0.2 but changes when I connect or
disconnect things to my computer and I wanted to make it static. The Getting
Started Guide suggests this is do-able but it seems I have to change things in
several places besides vars.xml like \sip_profiles\internal.xml and maybe other
places. Is there some general way to set my internal domain?
Thanks.
?
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Re: [Freeswitch-users] How do I set my FS internal ip address to a static value.
Ok, so is it $${local_ip_v4} that needs changing somewhere? I see stuff in
vars.xml for external address changes but not for internal ip address changes??
-Original Message-
From: Ken Rice kr...@suspicious.org
To: freeswitch-users@lists.freeswitch.org
Sent: Sun, 4 Jan 2009 12:48 pm
Subject: Re: [Freeswitch-users] How do I set my FS internal ip address to a
static value.
That is the DOMAIN not the IP... Hence the name is domain... If you look thru
vars.xml you’ll see other variables for the IP...
From: mszla...@aol.com
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Sun, 04 Jan 2009 15:14:27 -0500
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] How do I set my FS internal ip address to a
static value.
I believe there is a variable $${domain} that is used by FreeSwitch for it's
internal IP address. This address was 10.0.0.2 but changes when I connect or
disconnect things to my computer and I wanted to make it static. The Getting
Started Guide suggests this is do-able but it seems I have to change things in
several places besides vars.xml like \sip_profiles\internal.xml and maybe other
places. Is there some general way to set my internal domain?
Thanks.
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Re: [Freeswitch-users] How do I set my FS internal ip address to a static value.
I see the following in that file:
? X-PRE-PROCESS cmd=set data=bind_server_ip=auto/
and thought that's what might need changing to:
? X-PRE-PROCESS cmd=set data=bind_server_ip=10.0.0.3/
But your looking at the same variable I was and I'm guessing something else
might be in order like:
X-PRE-PROCESS cmd=set data=local_ip_v4=10.0.0.3/
I'll see if either of these work unless you have a different suggestion.
You and Brian also suggested a more specific approach in sip_profiles/*.xml and
input the sip-ip and rtp-ip for the sofia profile.
??? param name=sip-ip value=$${local_ip_v4}/? to param
name=sip-ip value=10.0.0.3/
??? param name=rtp-ip value=$${local_ip_v4}/?? to param
name=rtp-ip value=10.0.0.3/
What's the advantage to doing it in the sip_profiles\internal.xml file over the
general way in vars.xml?
?
?
-Original Message-
From: Jason White ja...@jasonjgw.net
To: freeswitch-users@lists.freeswitch.org
Sent: Sun, 4 Jan 2009 3:15 pm
Subject: Re: [Freeswitch-users] How do I set my FS internal ip address to a
static value.
mszla...@aol.com mszla...@aol.com wrote:
Ok, so is it? $${local_ip_v4} that needs changing somewhere? I see stuff in
vars.xml for external address changes but not for internal ip address
changes??
If you set $${local_ip_v4} in vars.xml it will determine which address
FreeSWITCH binds to, at least in the default configuration.
Addresses can be configured more flexibly in the SIP profiles.
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Re: [Freeswitch-users] How do I set my FS internal ip address to a static value.
Brian and Jason.
I tried changing $${local_ip_v4} to a static IP in the vars.xml and internal
and external sip_profiles files. This didn't work.
What I want to do is maybe more related to what local ip address my Windows
machine binds FreeSwitch to. It's currently associating it to 10.0.0.2 if I
change? $${local_ip_v4} to 10.0.0.3 then FS gets errors and sofia status
doesn't show any ip address associated with FS.
I'm guessing I need to get Windows always associating FS with a static IP
instead of a possibly changing value. I did this with a Linksys SPA3103 in it's
configuration menu so is there something analogous for FS and how do I do that?
Thanks.
-Original Message-
From: Brian West br...@freeswitch.org
To: freeswitch-users@lists.freeswitch.org
Sent: Sun, 4 Jan 2009 5:56 pm
Subject: Re: [Freeswitch-users] How do I set my FS internal ip address to a
static value.
I'm going to guess you don't have the latest configs. ?bind_server_ip is for
Dingaling ONLY at this point and is noted with the nice HUGE warning above the
setting. You can set the local_ip_v4 address but you're better off setting the
ip's in the profile.
/b
On Jan 4, 2009, at 6:23 PM, mszla...@aol.com wrote:
I see the following in that file:
? X-PRE-PROCESS cmd=set data=bind_server_ip=auto/
and thought that's what might need changing to:
? X-PRE-PROCESS cmd=set data=bind_server_ip=10.0.0.3/
But your looking at the same variable I was and I'm guessing something else
might be in order like:
X-PRE-PROCESS cmd=set data=local_ip_v4=10.0.0.3/
I'll see if either of these work unless you have a different suggestion.
You and Brian also suggested a more specific approach in sip_profiles/*.xml
and input the sip-ip and rtp-ip for the sofia profile.
??? param name=sip-ip value=$${local_ip_v4}/? to param
name=sip-ip value=10.0.0.3/
??? param name=rtp-ip value=$${local_ip_v4}/?? to param
name=rtp-ip value=10.0.0.3/
What's the advantage to doing it in the sip_profiles\internal.xml file over
the general way in vars.xml?
?
?
-Original Message-
From: Jason White ja...@jasonjgw.net
To: freeswitch-users@lists.freeswitch.org
Sent: Sun, 4 Jan 2009 3:15 pm
Subject: Re: [Freeswitch-users] How do I set my FS internal ip address to a
static value.
mszla...@aol.com mszla...@aol.com wrote:
Ok, so is it? $${local_ip_v4} that needs changing somewhere? I see stuff in
vars.xml for external address changes but not for internal ip address
changes??
If you set $${local_ip_v4} in vars.xml it will determine which address
FreeSWITCH binds to, at least in the default configuration.
Addresses can be configured more flexibly in the SIP profiles.
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Re: [Freeswitch-users] How do I set my FS internal ip address to a static value.
Brian, I did that and know it was every instance since I used Textpad's
replace function.
-Original Message-
From: Brian West br...@freeswitch.org
To: freeswitch-users@lists.freeswitch.org
Sent: Sun, 4 Jan 2009 10:13 pm
Subject: Re: [Freeswitch-users] How do I set my FS internal ip address to a
static value.
Well I did say open up sip_profiles/internal.xml or sip_profiles/external.xml
and every place it has $${local_ip_v4} REPLACE IT with the desired IP.
?Is?10.0.0.3 bound to your windows machine?
/b
On Jan 5, 2009, at 12:06 AM, mszla...@aol.com wrote:
I tried changing $${local_ip_v4} to a static IP in the vars.xml and internal
and external sip_profiles files. This didn't work.
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Re: [Freeswitch-users] How do I set my FS internal ip address to a static value.
Jason, I have no idea either but if I assign an IP address to the network
interface, say 10.0.0.3 then I may have another problem. I have a Voxeo
Prophecy ASR server being assigned to 10.0.0.2. Actually both FS and Prophecy
were being assigned by Windows to 10.0.0.2 but this was causing audio transfer
problems when I bridged FS to Prophecy. So I wanted to assign FS to something
else but I *believe* that Windows only uses one adaptor. What I need to do next
I don't know (create another adaptor) and probably I've never done.
-Original Message-
From: Jason White ja...@jasonjgw.net
To: freeswitch-users@lists.freeswitch.org
Sent: Sun, 4 Jan 2009 10:17 pm
Subject: Re: [Freeswitch-users] How do I set my FS internal ip address to a
static value.
mszla...@aol.com mszla...@aol.com wrote:
I tried changing $${local_ip_v4} to a static IP in the vars.xml and internal
and external sip_profiles files. This didn't work.
Make sure that one of your network interfaces - the one that you want
FreeSWITCH to use - is assigned that address first, otherwise there will be
nothing for FreeSWITCH to bind to when it starts.
I don't know much about Windows, so if you're running FreeSWITCH on Windows,
I'm sure there will be others on the list who are in a position to help.
If you're using Linux, just make sure that the address you want is associated
with an interface, e.g., using ifconfig, then start FreeSWITCH with the
pre-processor variable $${local_ip_v4} set appropriately.
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Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?
Hi Andy and Anthony. Thanks Anthony for elaborating more and I'll attempt using another IP on the same box as well. Also, Prophecy support has asked me first to put one application on a separate box and then get some wireshark data so I'll attempt that also. Andy, I didn't want to bother you given all those things you had to deal with. Welcome back. I explored the VMware idea before but was warned that it would not work well with an ASR. This advice came from the Trixbox forums, LumenVox, FreeSwitch and Voxeo. I understand that what I'm doing goes against the grain (i.e. voip) but frankly my target market really doesn't want anything to do with voip or even internet connectivity from their businesses. Plus there are other issues. I'll let you know how it goes. Happy holidays. Mark. -Original Message- From: Andrew Gilbert gilbertand...@me.com To: freeswitch-users@lists.freeswitch.org Sent: Mon, 22 Dec 2008 7:49 am Subject: 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 definite ly 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. 20 -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 app lication=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
Re: [Freeswitch-users] If Bypass Media works why won't Proxy Media work?
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 a dvis 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 – 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 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888
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 :-) -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 advised 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=2 0– 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 ___ 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] Crackling noise when bypassing media between endpoints.
When using bypass_media (aka. no_media) mode between an X-lite softphone and Prophacy ASR, I get intermittent crackiling background noise with the audio that I'm hearing. How do I get rid of this? ___ 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] Crackling noise when bypassing media between endpoints.
Man, I can't win with this one. I can bypass media between two endpoints with some static but what I really want FS to do is process the audio before it's passed on. However, getting FS involved is something I haven't had any success in with these two endpoints ... so far. Thanks for pointing out the noise source(s) with bypass ... makes sense given the name. -Original Message- From: Brian West br...@freeswitch.org To: freeswitch-users@lists.freeswitch.org Sent: Thu, 18 Dec 2008 10:31 am Subject: Re: [Freeswitch-users] Crackling noise when bypassing media between endpoints. Yes Chris you are right. FreeSWITCH isn't involved in the media at all. /b On Dec 18, 2008, at 12:14 PM, Chris wrote: I'm no expert, but I believe in media bypass mode freeswitch isn't handling media so it's not a fs fix, it would be the quality of connection for each of the originator/terminator, fs just directs each endpoint to set's up a point to point connection for RTP. Is this right? ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Help with routing sound locally through FS
Hi Mike, That does get the audio go between the softphone and the application (Voxeo's Prophecy ASR) around FreeSwitch but I would like the audio going through FreeSwitch. I plan to do something to it before passing it on. Support from Voxeo had this to say about the bypass media setting and if you could add some more insight that would be much appreciated. Since this is all on one Windows XP machine they can't get the info from the pcap file and are requesting I set up freeswitch on another machine which I will do. I thought you may have some more input. Mark, This is great news, it certainly confirms our suspicions that freeswitch was not forwarding media to Prophecy, or if so, it was doing it on a different port then we specified to be listening on. To address the lingering question in this thread, I don't believe we have a firm enough grasp on your deployment calls to understand whether free-switch need the RTP stream or not. If FreeSwitch is intended in your deployment to act as a front end for call routing to terminate calls to Prophecy then there is no need for it to listen to media. Of course, it will hold the SIP communication tether so that it remains aware of disconnect events, would be my assumption, I am sure freeswitch can verify this behavior. In order for us to understand why this config change is required will need a wireshark trace, and with your stacked approach to have both Prophecy and freeswitch on the same box makes this impossible. For troubleshooting, if you moved freeswitch to another server temporarily, this may offer some insight into this problem, with wireshark at our disposal. Hope this helps! -Original Message- From: Michael Jerris m...@jerris.com To: freeswitch-users@lists.freeswitch.org Sent: Tue, 16 Dec 2008 5:24 am Subject: Re: [Freeswitch-users] Help with routing sound locally through FS If its all local you can also just use: http://wiki.freeswitch.org/wiki/Bypass_Media If your still trying to figure it out it could be any number of things, but most relating to misconfigured endpoints or freeswitch, take a look at the sip trace and make sure everything is using the right ip addresses instead of using internal when they should be external or the other way around. Mike On Dec 16, 2008, at 5:02 AM, mszla...@aol.com wrote: I'm making a call internally from a soft phone to an extension that is suppose to bridge the call internally to another application on the same computer. The applications logs indicate that a connection was made but sound is not being passed back from the application through freeswitch to the softphone. There maybe an issue with rtp timing and associated ports but I'm very new at diagnosing this and fixing the problem. I've attached both a copy of the FS log and an associated pcap file. It's all on Windows XP. Could someone please take a look. Thanks. Listen to 350+ music, sports, news radio stations – including songs for the holidays – FREE while you browse. Start Listening Now! free.zip___ 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] Confirmation checking on VAD is needed.
I need confirmation on whether VAD is disabled in the FS default configuration. I've looked at conf\sip_profiles\internal.xml and the relevant section was the following which seems to indicate that VAD is off by default since I don't see VAD value=none. ??? !-- VAD choose one (out is a good choice); -- ??? !-- param name=vad value=in/ -- ??? !-- param name=vad value=out/ -- ??? !-- param name=vad value=both/ -- ??? !--param name=alias value=sip:10.0.1.251:/-- ??? !--all inbound reg will look in this domain for the users -- ??? !--param name=force-register-domain value=cluecon.com/-- ??? !-- disable register and transfer which may be undesirable in a public switch -- ??? !--param name=disable-transfer value=true/-- ??? !--param name=disable-register value=true/-- However, is there a way to confirm this in the FS console? ___ 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] Audio routing problem between FS and Voxeo
I’m trying to route calls from X-lite -- FS (Nov. 6 2008 svn) -- Voceo (Prophecy) to use Voxeo’s ASR instead of FS’s built in PocketSphinx/ASR. All these applications reside on the same computer/OS (Win XP). I have at Netgear wifi router that connects a laptop to my desktop in case that matters but I’m not using the laptop for any of this. I’ve set up an extension to bridge calls to Voxeo. Here is the entry in file conf\dialplan\default.xml: extension name=Doctors Office condition field=destination_number expression=^2007$ action application=info/ action application=bridge data=sofia/internal/sip:[EMAIL PROTECTED]:5068/ /condition /extension I hear one ring then a hang up. No errors in the FS console and Voxeo’s logs ramp up when I dial the 2007 extension on X-lite but I do not get any audio from the dialogue script in Voxeo’s Prophecy ASR called “Doctorsoffice.” Any ideas? Thanks. 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
Re: [Freeswitch-users] Audio routing problem between FS and Voxeo
Yes but someone else I'm in contact with set up FS a couple days ago and is having the same problems. Brian should I still update today? -Original Message- From: Brian West [EMAIL PROTECTED] To: freeswitch-users@lists.freeswitch.org Sent: Wed, 10 Dec 2008 10:45 am Subject: Re: [Freeswitch-users] Audio routing problem between FS and Voxeo You're a month behind.. I highly recommend you update. Chances are this has already been fixed. /b On Dec 10, 2008, at 12:39 PM, [EMAIL PROTECTED] wrote: I’m trying to route calls from X-lite -- FS (Nov. 6 2008 svn) -- Voceo (Prophecy) to use Voxeo’s ASR instead of FS’s built in PocketSphinx/ASR. All these applications reside on the same computer/OS (Win XP). I have at Netgear wifi=2 0router that connects a laptop to my desktop in case that matters but I’m not using the laptop for any of this. I’ve set up an extension to bridge calls to Voxeo. Here is the entry in file conf\dialplan\default.xml: extension name=Doctors Office condition field=destination_number expression=^2007$ action application=info/ action application=bridge data=sofia/internal/sip:[EMAIL PROTECTED]:5068/ /condition /extension I20hear one ring then a hang up. No errors in the FS console and Voxeo’s logs ramp up when I dial the 2007 extension on X-lite but I do not get any audio from the dialogue script in Voxeo’s Prophecy ASR called “Doctorsoffice.” Any ideas? Thanks. Mark. 0A 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 ___ 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] Audio routing problem between FS and Voxeo
It was someone from Voxeo support. I think John was the main person helping me with this. I updated but things got worse all over. I now can't run other extensions Gino's pizza or some db stuff Got these errors: 2008-12-10 11:40:23 [ERR] mod_spidermonkey.c:928 sm_load_file() Error Loading module C:\Source\freeswitch-snapshot\Debug\mod\mod_spidermonkey_teletone.dll **The specified module could not be found.? ** 2008-12-10 11:40:23 [CONSOLE] mod_spidermonkey.c:944 sm_load_file() Successfully Loaded [C:\Source\freeswitch-snapshot\Debug\mod\mod_spidermonkey_core_db.dll] 2008-12-10 11:40:23 [CONSOLE] mod_spidermonkey.c:944 sm_load_file() Successfully Loaded [C:\Source\freeswitch-snapshot\Debug\mod\mod_spidermonkey_socket.dll] 2008-12-10 11:40:23 [CONSOLE] switch_loadable_module.c:789 switch_loadable_module_load_file() Successfully Loaded [mod_spidermonkey] 2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:238 switch_loadable_module_process() Adding Application 'javascript' 2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:258 switch_loadable_module_process() Adding API Function 'jsrun' 2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:258 switch_loadable_module_process() Adding API Function 'jsapi' 2008-12-10 11:40:23 [CONSOLE] switch_loadable_module.c:789 switch_loadable_module_load_file() Successfully Loaded [mod_lua] 2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:238 switch_loadable_module_process() Adding Application 'lua' 2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:258 switch_loadable_module_process() Adding API Function 'luarun' 2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:258 switch_loadable_module_process() Adding API Function 'lua' 2008-12-10 11:40:23 [CONSOLE] switch_loadable_module.c:789 switch_loadable_module_load_file() Successfully Loaded [mod_say_en] 2008-12-10 11:40:23 [NOTICE] switch_loadable_module.c:371 switch_loadable_module_process() Adding Say interface 'en' 2008-12-10 11:40:23 [CONSOLE] switch_loadable_module.c:118 switch_loadable_module_runtime() Starting runtime thread for CORE_SOFTTIMER_MODULE 2008-12-10 11:40:23 [CONSOLE] switch_loadable_module.c:118 switch_loadable_module_runtime() Starting runtime thread for mod_event_socket 2008-12-10 11:40:23 [CONSOLE] switch_core.c:862 switch_load_network_lists() Created ip list dl-candidates default (allow) 2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 10.0.0.0/8 (deny) to list dl-candidates 2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 172.16.0.0/12 (deny) to list dl-candidates 2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 192.168.0.0/16 (deny) to list dl-candidates 2008-12-10 11:40:23 [CONSOLE] switch_core.c:862 switch_load_network_lists() Created ip list rfc1918 default (deny) 2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 10.0.0.0/8 (allow) to list rfc1918 2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 172.16.0.0/12 (allow) to list rfc1918 2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 192.168.0.0/16 (allow) to list rfc1918 2008-12-10 11:40:23 [CONSOLE] switch_core.c:862 switch_load_network_lists() Created ip list lan default (allow) 2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 192.168.42.0/24 (deny) to list lan 2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 192.168.42.42/32 (allow) to list lan 2008-12-10 11:40:23 [CONSOLE] switch_core.c:862 switch_load_network_lists() Created ip list strict default (deny) 2008-12-10 11:40:23 [NOTICE] switch_core.c:919 switch_load_network_lists() Adding 208.102.123.124/32 (allow) to list strict 2008-12-10 11:40:23 [CONSOLE] switch_core.c:862 switch_load_network_lists() Created ip list domains default (deny) 2008-12-10 11:40:23 [NOTICE] switch_core.c:907 switch_load_network_lists() Adding 1.2.3.4/24 (allow) [EMAIL PROTECTED] to list domains 2008-12-10 11:40:23 [CONSOLE] switch_core.c:1258 switch_core_init_and_modload() FreeSWITCH Version 1.0.trunk (10171M) Started. Crash Protection [Disabled] Max Sessions[1000] Session Rate[30] SQL [Enabled] [EMAIL PROTECTED] 2008-12-10 11:40:24 [ERR] sofia.c:543 sofia_profile_thread_run() Error Creating SIP UA for profile: internal-ipv6 2008-12-10 11:40:45 [NOTICE] switch_channel.c:553 switch_channel_set_name() New Channel sofia/internal/[EMAIL PROTECTED] [88d1aaf9-a625-444e-883d-c5ac6eeac30e] 2008-12-10 11:40:45 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing FreeSwitch-2006 in context default 2008-12-10 11:40:45 [NOTICE] mod_spidermonkey.c:2034 session_answer() Channel [sofia/internal/[EMAIL PROTECTED] has been answered 2008-12-10 11:40:48 [ERR] switch_core_asr.c:57 switch_core_asr_open() Invalid ASR module [pocketsphinx]! 2008-12-10 11:40:48 [NOTICE] switch_ivr_async.c:1845 switch_ivr_detect_speech()
Re: [Freeswitch-users] Sounds for pending 1.0.2/Hardware
How do I donate? -Original Message- From: Brian West [EMAIL PROTECTED] To: freeswitch-users@lists.freeswitch.org Cc: [EMAIL PROTECTED] Sent: Wed, 10 Dec 2008 12:27 pm Subject: [Freeswitch-users] Sounds for pending 1.0.2/Hardware FreeSWITCHers, I'm looking for donations for the next batch of sound files we need to have done for the up coming 1.0.2 release. I have had others pitch in some money in the past and I thank everyone for doing so. I hope everyone can come together and help me raise about $200 to pay for this batch of prompts. I also would like to thank Bandwidth.com and Teliax for their support of the FreeSWITCH project. Are you ready for 1.0.2? Go download SVN Trunk and beat it up for us... If you wish to donate please paypal [EMAIL PROTECTED] that'll help out! Happy Holidays, Brian West FreeSWITCH.org PS: If you know of any sound files we need let me know. ___ 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] Audio routing problem between FS and Voxeo
Yup my bad. But I'm still getting this error: 2008-12-10 13:18:05 [ERR] mod_spidermonkey.c:928 sm_load_file() Error Loading module C:\Source\freeswitch-snapshot\Debug\mod\mod_spidermonkey_teletone.dll **The specified module could not be found.? ** It doesn't look like it was put in this latest snapshot. I could use that dll from my older snapshot, has it been changed since then? I'm still having the same problem with no audio from Voxeo. Mark. -Original Message- From: Brian West [EMAIL PROTECTED] To: freeswitch-users@lists.freeswitch.org Sent: Wed, 10 Dec 2008 11:59 am Subject: Re: [Freeswitch-users] Audio routing problem between FS and Voxeo Looks like mod_pocketsphinx and mod_openmrcp isn't loaded. /b On Dec 10, 2008, at 1:51 PM, [EMAIL PROTECTED] wrote: It was someone from Voxeo support. I think John was the main person helping me with this. I updated but things got worse all over. I now can't run other extensions Gino's pizza or some db stuff 2008-12-10 11:40:48 [ERR] switch_core_asr.c:57 switch_core_asr_open() Invalid ASR module [pocketsphinx]! ?2008-12-10 11:41:23 [ERR] switch_core_speech.c:60 switch_core_speech_open() Invalid speech module [openmrcp]! 2008-12-10 11:41:23 [ERR] mod_spidermonkey.c:1859 init_speech_engine() Invalid TTS module! 2008-12-10 11:41:23 [ERR] inline:1 mod_spidermonkey()? Cannot allocate speech engine! 2008-12-10 11:41:23 [NOTICE] switch_core_state_machine.c:160 switch_core_standard_on_execute() Hangup sofia/internal/[EMAIL PROTECTED] [CS_EXECUTE] [NORMAL_CLEARING] 2008-12-10 11:41:23 [NOTICE] switch_core_session.c:878 switch_core_session_thread() Session 2 (sofia/internal/[EMAIL PROTECTED]) Ended 2008-12-10 11:41:23 [NOTICE] switch_core_session.c:880 switch_core_session_thread() Close Channel sofia/internal/[EMAIL PROTECTED] [CS_HANGUP] = ___ 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] Audio routing problem between FS and Voxeo
Brian, I'm still having the same audio problem when bridging/routing to Voxeo using the latest snapshot. Help! Also, it doesn't look like mod_spidermonkey_teletone.dll was put into this latest snapshot. Has it been changed since November? I could use that older version of this dll. 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
Re: [Freeswitch-users] Audio routing problem between FS and Voxeo
I only glanced at the compilers output when it ended and it reported no no errors but I did not look at the warnings along the way. Have things been modified since so spider monkey compiles? More importantly, do you have any ideas as what is going on with my audio problem. I can't attach a wireshark .pcap file since it's to big for your list and my email gets rejected. -Original Message- From: Carlos Talbot [EMAIL PROTECTED] To: freeswitch-users@lists.freeswitch.org freeswitch-users@lists.freeswitch.org Cc: freeswitch-users@lists.freeswitch.org freeswitch-users@lists.freeswitch.org Sent: Wed, 10 Dec 2008 2:57 pm Subject: Re: [Freeswitch-users] Audio routing problem between FS and Voxeo There was a typecast warning that prevented spidermoneky from compiling in a recent svn. Did you check to see if it compiled? Sent from my iPhone On Dec 10, 2008, at 3:24 PM, [EMAIL PROTECTED] wrote: Yup my bad. But I'm still getting this error: 2008-12-10 13:18:05 [ERR] mod_spidermonkey.c:928 sm_load_file() Error Loading module C:\Source\freeswitch-snapshot\Debug\mod\mod_spidermonkey_teletone.dll **The specified module could not be found. ** It doesn't look like it was put in this latest snapshot. I could use that dll from my older snapshot, has it been changed since then? I'm still having the same problem with no audio from Voxeo. Mark. -Original Message- From: Brian West [EMAIL PROTECTED] witch.org To: freeswitch-users@lists.freeswitch.org Sent: Wed, 10 Dec 2008 11:59 am Subject: Re: [Freeswitch-users] Audio routing problem between FS and Voxeo Looks like mod_pocketsphinx and mod_openmrcp isn't loaded. /b On Dec 10, 2008, at 1:51 PM, [EMAIL PROTECTED] wrote: It was someone from Voxeo support. I think John was the main person helping me with this. I updated but things got worse all over. I now can't run other extensions Gino's pizza or some db stuff 2008-12-10 11:40:48 [ERR] switch_core_asr.c:57 switch_core_asr_open() Invalid ASR module [pocketsphinx]! 2008-12-10 11:41:23 [ERR] switch_core_speech.c:60 switch_core_speech_open() Invalid speech module [openmrcp]! 2008-12-10 11:41:23 [ERR] mod_spidermonkey.c:1859 init_speech_engine() Invalid TTS module! 2008-12-10 11:41:23 [ERR] inline:1 mod_spidermonkey() Cannot allocate speech engine! 2008-12-10 11:41:23 [NOTICE] switch_core_state_machine.c:160 switch_core_standard_on_execute() Hangup sofia/internal/[EMAIL PROTECTED] [CS_EXECUTE] [NORMAL_CLEARING] 2008-12-10 11:41:23 [NOTICE] switch_core_session.c:878 switch_core_session_thread() Session 2 (sofia/internal/[EMAIL PROTECTED]) Ended 2008-12-10 11:41:23 [NOTICE] switch_core_session.c:880 switch_core_session_thread() Close Channel sofia/internal/[EMAIL PROTECTED] [CS_HANGUP] = ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitc h-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 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 ___ 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] [Freeswitch-dev] Sounds for pending 1.0.2/Hardware
Good enough will do to get my cash. It will be on the way once my paypal account is confirmed in a few days. -Original Message- From: Brian West [EMAIL PROTECTED] To: freeswitch-users@lists.freeswitch.org Sent: Wed, 10 Dec 2008 5:56 pm Subject: Re: [Freeswitch-users] [Freeswitch-dev] Sounds for pending 1.0.2/Hardware Thank you it really helps. I want to make sure the 1.0.2 release is the best release ever! /b On Dec 10, 2008, at 7:51 PM, Angel Carpintero wrote: I'm in too . Brian hope you got money i sent, a pleasure to contribute. ___ 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
Hi Anthony, Oh! OK. So is this module totally broken. I say this because I can't seem to get it to work at all with the example in that Mod_openmrcp wiki page but I thought it might because I'm not be using the right Cepstral software (freetrial download versus the paided for SDK) or that I'm not using the right port numbers or something else I didn't do. I used TcpView to look at local port associated with my Cepstral software and changed a few things but still nothing. I changed the loglevel setting to 7 in the wiki's example but I don't see the kind of output on the console that I would expect for debug mode. Thanks. Mark. -Original Message- From: Anthony Minessale [EMAIL PROTECTED] To: freeswitch-users@lists.freeswitch.org Sent: Mon, 1 Dec 2008 9:37 am Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP the guy who made mod_openmrcp has stopped development and is now making a new library called unimrcp it will take some time to create a new module and remove the now unsupported openmrcp. On Fri, Nov 28, 2008 at 12:15 PM, [EMAIL PROTECTED] wrote: I'm getting the following errors when trying to run the example in the wiki: http://wiki.freeswitch.org/wiki/Mod_openmrcp 2008-11-28 09:59:54 [DEBUG] switch_core_session.c:435 switch_core_session_receive_message() Send signal sofia/internal/[EMAIL PROTECTED] [BREAK] 2008-11-28 09:59:54 [DEBUG] sofia.c:2269 sofia_handle_sip_i_state() Channel sofia/internal/[EMAIL PROTECTED] entering state [completed] 2008-11-28 09:59:54 [NOTICE] mod_spidermonkey.c:2034 session_answer() Channel [sofia/internal/[EMAIL PROTECTED] has been answered 2008-11-28 09:59:54 [DEBUG] mod_spidermonkey.c:1851 init_speech_engine() Raw Codec Activation Success [EMAIL PROTECTED] 1 channel 20ms 2008-11-28 09:59:54 [DEBUG] mod_openmrcp.c:634 openmrcp_tts_open() Create Synthesizer Channel 2008-11-28 09:59:54 [DEBUG] sofia.c:2269 sofia_handle_sip_i_state() Channel sofia/internal/[EMAIL PROTECTED] entering state [ready] 2008-11-28 09:59:59 [ERR] mod_openmrcp.c:643 openmrcp_tts_open() No response from client stack 2008-11-28 09:59:59 [ERR] mod_openmrcp.c:647 openmrcp_tts_open() No synthesizer channel available 2008-11-28 09:59:59 [ERR] mod_spidermonkey.c:1859 init_speech_engine() Invalid TTS module! 2008-11-28 09:59:59 [ERR] inline:1 mod_spidermonkey()? Cannot allocate speech engine! 2008-11-28 09:59:59 [NOTICE] switch_core_state_machine.c:160 switch_core_standard_on_execute() Hangup sofia/internal/[EMAIL PROTECTED] [CS_EXECUTE] [NORMAL_CLEARING] 2008-11-28 09:59:59 [DEBUG] switch_channel.c:1449 switch_channel_perform_hangup() Send signal sofia/internal/[EMAIL PROTECTED] [KILL] 2008-11-28 09:59:59 [DEBUG] switch_core_session.c:727 switch_core_session_signal_state_change() Send signal sofia/internal/[EMAIL PROTECTED] [BREAK] 2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:432 switch_core_session_run() (sofia/internal/[EMAIL PROTECTED]) State EXECUTE going to sleep 2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:367 switch_core_session_run() (sofia/internal/[EMAIL PROTECTED]) Running State Change CS_HANGUP 2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:395 switch_core_session_run() (sofia/internal/[EMAIL PROTECTED]) State HANGUP 2008-11-28 09:59:59 [DEBUG] mod_sofia.c:276 sofia_on_hangup() Channel sofia/internal/[EMAIL PROTECTED] hanging up, cause: NORMAL_CLEARING 2008-11-28 09:59:59 [DEBUG] mod_sofia.c:333 sofia_on_hangup() Sending BYE to sofia/internal/[EMAIL PROTECTED] 2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() sofia/internal/[EMAIL PROTECTED] Standard HANGUP, cause: NORMAL_CLEARING 2008-11-28 09:59:59 [DEBUG] switch_core_state_machine.c:395 switch_core_session_run() (sofia/internal/[EMAIL PROTECTED]) State HANGUP going to sleep 2008-11-28 09:59:59 [DEBUG] switch_core_session.c:860 switch_core_session_thread() Session 1 (sofia/internal/[EMAIL PROTECTED]) Locked, Waiting on external entities 2008-11-28 09:59:59 [NOTICE] switch_core_session.c:878 switch_core_session_thread() Session 1 (sofia/internal/[EMAIL PROTECTED]) Ended 2008-11-28 09:59:59 [NOTICE] switch_core_session.c:880 switch_core_session_thread() Close Channel sofia/internal/[EMAIL PROTECTED] [CS_HANGUP] 2008-11-28 10:00:26 [DEBUG] mod_openmrcp.c:167 openmrcp_on_session_terminate() on_session_terminate called I believe I followed the instructions correctly but I can't get openmrcp to connect with Cepstrals TTS. Tis the season to save your money! Get the new AOL Holiday Toolbar for money saving offers and gift ideas. ___ 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
Re: [Freeswitch-users] Problems with Mod_openMRCP
MikeJ, if openMRCP isn't totally broken then would you mind helping me get the example in Mod_openMRCP working or something like it since I don't know what the heck I'm doing wrong. I can meet you now over at the IRC channel for Freeswitch users if you like. Thanks. -Original Message- From: Michael Jerris [EMAIL PROTECTED] To: freeswitch-users@lists.freeswitch.org Sent: Mon, 1 Dec 2008 10:30 am Subject: Re: [Freeswitch-users] Problems with Mod_openMRCP I would not say it is totally broken, it is known to work in quite a few places, but we are unlikely to be doing any new fixes in it. Mike On Dec 1, 2008, at 1:19 PM, [EMAIL PROTECTED] wrote: Hi Anthony, Oh! OK. So is this module totally broken. I say this because I can't seem to get it to work at all with the example in that Mod_openmrcp wiki page but I thought it might because I'm not be using the right Cepstral software (freetrial download versus the paided for SDK) or that I'm not using the right port numbers or something else I didn't do. I used TcpView to look at local port associated with my Cepstral software and changed a few things but still nothing. I changed the loglevel setting to 7 in the wiki's example but I don't see the kind of output on the console that I would expect for debug mode. Thanks. 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 ___ 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
