April 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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

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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

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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

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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

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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....

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Freeswitch 1.67. is out
by Gregor Nanger 05 Apr '16

05 Apr '16

This could be silly question for someone, but anyway. We all have to learn something new . Can someone please explain me how code is managed in such large projects. I saw that FS 1.6.7 is out and has a lot of mod_verto features added. Are those features already in latest master branch? Is master 1.7 version?

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Re: [Freeswitch-users] IMT Codec Enforcement
by Ahmed habiba 05 Apr '16

05 Apr '16

Your kind feedback will be appreciated. > > > From: Ahmed habiba <ahabiba(a)gmail.com> > Subject: [Freeswitch-users] IMT Codec Enforcement > Date: April 4, 2016 at 10:37:46 PM GMT+3 > To: freeswitch-users(a)lists.freeswitch.org > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > Dears, > > kindly I have the below scenario: > > [SoftPhone1]<——Channel1——>[FreeSwitch1]<——Channel2——>[FreeSwitch2]<——Chanel3——>[SoftPhone2] > > Channel1 codec G722 > > Channel3 codec G722 > > I want to enforce Channel 3 between FreeSwitch instances to be GSM, I change the Dial plane of freeSwitch1 to and the same for FS2 > > <action application="set" data=“absolute_codec_string=GSM"/> > > <action application="bridge" data="sofia/internal/$1(a)10.8.0.22 <mailto:sofia/internal/$1@10.8.0.22>"/> > > However when I run call from FS2 to FS1, FS1 reply with the below putting “a=rtpmap:106 opus/48000/2” as the only available codec as below, your kind help will be appreciated > > > > recv 1275 bytes from udp/[10.8.0.22]:5060 at 15:22:35.560845: > ------------------------------------------------------------------------ > SIP/2.0 183 Session Progress > Via: SIP/2.0/UDP 10.8.0.29;rport=5060;branch=z9hG4bKSr1v2mgcctX5H > From: "Extension 1000" <sip:1000@10.8.0.29 <sip:1000@10.8.0.29>>;tag=U82NjmSa9S9Kg > To: <sip:1000@10.8.0.22 <sip:1000@10.8.0.22>>;tag=g0eNeprr8e4Fg > Call-ID: 6d5dbd7f-753d-1234-d78c-bc305bd3840c > CSeq: 89562813 INVITE > Contact: <sip:1000@10.8.0.22:5060;transport=udp <sip:1000@10.8.0.22:5060;transport=udp>> > User-Agent: FreeSWITCH-mod_sofia/1.6.6+git~20160111T201612Z~d2d0b3283a~64bit > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, path, replaces > Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 261 > X-FS-Display-Name: 1000 > X-FS-Display-Number: sip:1000@10.8.0.22 <sip:1000@10.8.0.22> > X-FS-Support: update_display,send_info > Remote-Party-ID: "1000" <sip:1000@10.8.0.22 <sip:1000@10.8.0.22>>;party=calling;privacy=off;screen=no > > v=0 > o=FreeSWITCH 1459771930 1459771931 IN IP4 10.8.0.22 > s=FreeSWITCH > c=IN IP4 10.8.0.22 > t=0 0 > m=audio 25836 RTP/AVP 106 101 > a=rtpmap:106 opus/48000/2 > a=fmtp:106 useinbandfec=1; minptime=20 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > > Thanks, > > Ahmed Habiba. >

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Thursday FreeSWITCH Bug Hunt
by Ítalo Rossi 05 Apr '16

05 Apr '16

FreeSWITCHers, Join us thursday 2PM CST for the Thursday FreeSWITCH Bug Hunt! Where? [conference.freeswitch.org/vc/#/?autocall=888](https://conference.frees witch.org/vc/#/?autocall=888 "https://conference.freeswitch.org/vc/#/?autocall=888" ) Chat? <https://hipchat.freeswitch.org/gYzPMg3Oc> What? FreeSWITCH Bug Hunt, Jira Reviews, and General FS Support! Help us help you, Join the Bug Hunt! Ítalo Rossi italo(a)freeswitch.org IRC chat.freenode.net #freeswitch #freeswitch-dev Bugs? https://freeswitch.org/jira Docs? https://freeswitch.org/jira Chat? https://hipchat.freeswitch.org/gUdAgy0m6

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Using lib sofia-sip outside of FreeSWITCH
by Ehsan Minachi 05 Apr '16

05 Apr '16

Hello everyone, I know FreeSWITCH maintains Sofia library internally. I'm building a distributed presence server using sofia, however I'm having hard time linking it against current supplied version of sofia lib under libs directory. I would very much prefer to link my app against the FreeSWITCH maintained lib vs. the one found over source forge. Any hints on how I should call ./configure on libs/sofia-sip is highly appreciated. Thank you very much. Best regards, - Ehsan

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