- FreeSWITCH-users - FreeSWITCH Mailing Lists

Verto firefox
by Fernando Villarroel 06 Apr '16

06 Apr '16

1 0

Record and live stream WAV to HTTP server
by Vincent Gire 06 Apr '16

06 Apr '16

Hi all, Thank you to all contributing to FreeSWITCH ! I'm working on a IVR project where logic is implemented on a HTTP server. We are leaving Twilio because we now need to record and live stream the session to the HTTP server in WAV format (chunked transfer encoding). *mod_httapi* looks great (HT TAPI very similar to Twilio's) but it seems that the records are first saved to disk before there are sent to the server as chunked data. We need the transfer to start as soon as the recording starts. *mod_shout* does start the request almost as the records starts but it does not support WAV file and shout:// is not exactly a HTTP request (SOURCE method instead of PUT). Is there a way to use these modules to achieve our goal ? If not, we are willing to author a specific module or rather contribute to the existing ones. We've identified two approaches: 1. From *mod_httapi* Modify mod_httapi to directly stream the record instead of completely saving it to disk before the HTTP chunked transfer starts. This seems the most logical but with more than 3000 lines, mod_httapi does not seem to be the easiest module to build upon for newcomers! 2. From *mod_shout* 1. Modify libshoot to replace the custom SOURCE method with standard HTTP PUT method 2. Modify mod_shout to support wav files 3. Implement our IVR in script (javascript/lua) What do you think ? Thank you for your help. Cheers, Vincent

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Error Opening DB
by Agustí Ubalde 06 Apr '16

06 Apr '16

Hello, In the console Freeswitch I have observed the following error: 2016-04-06 11:35:08.864016 [ERR] switch_core_db.c:108 SQL ERR [unsupported file format] 2016-04-06 11:35:08.864016 [ERR] switch_core_db.c:223 SQL ERR [unsupported file format] 2016-04-06 11:35:08.864016 [CRIT] switch_core_sqldb.c:507 Failure to connect to CORE_DB voicemail_default! 2016-04-06 11:35:08.864016 [ERR] mod_voicemail.c:297 Error Opening DB Anyone know how to fix it? Is possible to disable the use of internal databases? Thanks and regards, Agustí *PRESENCE TECHNOLOGY* *Agustí Ubalde Bellot* Chief Developer C/ Comte Urgell 240 3A Barcelona 08036 aubalde(a)presenceco.com Ph: +34 93 10 10 300 Fx: +34 93 10 10 333 *www.presenceco.com* <http://www.presenceco.com/> *Follow us on:* *[image: tw]* <http://twitter.com/presencetech>*[image: yt]* <http://www.youtube.com/PresenceTechnology>*[image: in]* <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]* <http://www.slideshare.net/PresenceTech>*[image: fb]* <http://www.facebook.com/presencetech> For additional information, please visit our website *www.presenceco.com* <http://www.presenceco.com/> -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com <http://www.presenceco.com/>*

2 1

480 Temporarily Unavailable[MANDATORY_IE_MISSING]
by Rutu Patel 06 Apr '16

06 Apr '16

Hi, We are sending call to asterisk PBX but we are getting 480 Temporarily Unavailable immediately. from freeswitch server. Call flow is as below: User => opensips => FS => PBX, but call not sending out from FS and drop by FS. Here I am attaching both sip logs and freeswitch logs. Can anyone please help to sort out the issue. -- Thanks, Rutu Patel <http://www.inextrix.com>

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comand convert
by amani mansour 05 Apr '16

05 Apr '16

Hi , i need to convert this comand shell to python : sudo tshark -n -r $pcap -Y "rtp && rtp.ssrc == $ssrc" -T fields -e rtp.payload | sed "s/:/ /g" | perl -ne 's/([0-9a-f]{2})/print chr hex $1/gie' >> $pcap.$ssrc.raw i do this : p2 = sub.Popen(('sudo', 'tshark','-n','-r',pcap,'-Y','rtp and rtp.ssrc=='+pa[n],'-T','fields','-e', 'rtp.payload','|',' sed','s','/',':','/',' /','g' ,'|',' perl',' -ne', 's','/,','([0-9a-f]{2}),','/','print',' chr',' hex',law,'/gie',' >>',pcap+'.'+pa[n]+'.raw'),stdout=sub.PIPE) is there some one who can help me please ? best regards amani

1 0

Re: [Freeswitch-users] Better results from mod_vmd
by Abaci B 05 Apr '16

05 Apr '16

I just tried mod_avmd and mod_vmd and seem to have the same issue, my question is if someone has this working with Panasonic answering machine and Verizon Wireless (as well as other US wireless carriers) voicemail, what change if any should I apply to get it working. Thanks for the help On Thu, Aug 20, 2009 at 8:36 PM, Matthew Fong <mattdfong(a)gmail.com> wrote: > I changed > > /*! Minimum time for a beep. */ > #define MIN_TIME 8000 > to 6500 and it seemed to work, but I'm not sure how many false positives I > will get in a real-world environment. at 4000 it fired the event like 5 > times in a session, but 6500 only once. Do you think I should expect a lot > of false positives after changing this value? > > --matt > http://www.hellohunter.com > > On Thu, Aug 20, 2009 at 4:54 PM, Eric des Courtis < > eric.des.courtis(a)gmail.com> wrote: > >> Matt, >> >> As is mod_vmd will not detect tones shorter then 138ms. However I >> could get that value down to ~30ms at best by making a few >> modifications to the algorithm. >> >> Cheers. >> >> Eric des Courtis >> >> >> On Thu, Aug 20, 2009 at 7:51 PM, Eric des >> Courtis<eric.des.courtis(a)gmail.com> wrote: >> > Matt, >> > >> > For your information the tones you gave me are exactly 738Hz. If you >> > want to try that tone detection thing. >> > >> > Cheers. >> > >> > Eric des Courtis >> > >> > On Thu, Aug 20, 2009 at 2:20 PM, Michael Collins<msc(a)freeswitch.org> >> wrote: >> >> >> >> >> >> On Thu, Aug 20, 2009 at 11:06 AM, Steve Underwood <steveu(a)coppice.org> >> >> wrote: >> >>> >> >>> On 08/20/2009 05:22 AM, Michael Collins wrote: >> >>> > >> >>> > There is no noise on those 3 beeps. In fact, for something >> that's >> >>> > been >> >>> > through ulaw/alaw compression those beeps are very clean. They >> are >> >>> > quite >> >>> > short, though. >> >>> > >> >>> > >> >>> > Heck yeah they're short! Steve, in your experience is there a >> >>> > practical way to detect a beep that short without chewing up system >> >>> > resources or having lots of false positives? >> >>> > -MC >> >>> > >> >>> The tone samples I just looked at are about 130ms long. The problem is >> >>> the detector is trying to be a very open ended detector of anything >> >>> narrowband enough to be a single tone, and of any duration beyond some >> >>> small minimum. Its difficult to make such a thing voice immune unless >> >>> you can also count on a very large signal to noise ratio. With a >> digital >> >>> trunk you can probably rely on a large SNR, but what happens when >> people >> >>> use analogue lines? There is a reason why DTMF detectors try hard to >> >>> work down to about 10dB SNR. :-) >> >>> >> >>> Steve >> >> >> >> Thanks for the lesson uncle Steve! I'm guessing that the OP will need >> a new >> >> strategy. Possibly waiting for silence? Not sure what's the best way >> to go >> >> but I'm interested in hearing if someone has a solution. >> >> >> >> -MC >> >> >> >> >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users(a)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(a)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(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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Re: [Freeswitch-users] Record and live stream WAV to HTTP server
by jelena＠misticnabica.hr 05 Apr '16

05 Apr '16

1 0

Specific ring-device for verto communicator (or webrtc in general)
by François 05 Apr '16

05 Apr '16

Hi, Would it be possible to for a webrtc/verto application to use a different audio device for "ringing" purposes (like most softphones do)? Thanks, F

3 3

Format_cdr modul
by Gregor Nanger 05 Apr '16

05 Apr '16

Hi! I would like to use format_cdr module to post json cdr. I can see there is format_cdr.conf file, but module is not defined in modules.conf. How can I enable this module? Best regards, Gergor

3 3

FS not see invite
by Yuriy Gorlichenko 05 Apr '16

05 Apr '16

Hello. I try to make a call through another SIP instalse to freeswitch. I added extension and use FS as proxy to rpovider. I successfully registered on FS as user and try to make a call. I see at FS cli that invite arrives and FS answers 407, next my instanse Sends ACK and thn INVITE with a digest auth. I see thins invite at the TCPDUMP on the FS machine but not see it at the FS Firewall is disabled for tests....

3 6