- FreeSWITCH-dev - FreeSWITCH Mailing Lists

Friday Free For All Activate! Get in here
by Ken Rice 25 Oct '13

25 Oct '13

sip:888@conference.freeswitch.org or http://fs0.us/call888 for other access info. Todays topic?? What ever we want it to be! Come on and find out what its all about! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch

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New "mod_dahdi_timer" - does the community want this?
by Bob Hartwig 23 Oct '13

23 Oct '13

I ran across a problem with a Dahdi-based FreeTDM system a few weeks ago. Because the timer for mod_conference was not synchronized to the T1 clock source, there was significant periodic on-conference noise. The problem was not that the "soft" timer was inaccurate; on the contrary, it was a very accurate 1 mS timer in a system that had a pretty inaccurate T1 clock coming in. My solution was to create a new "mod_dahdi_timer", which derives the ~1 mS timer from 8 ticks of the 8 kHz /dev/dahdi/timer device. Then a one-line change to mod_conference.c to change "soft" to "dahdi", and the conference sounds perfectly clean. Is this something that the FS community wants? It's pretty specific - probably only of interest to users of Dahdi-based FreeTDM systems, but for them, it may be very useful. If so, how do I go about contributing it? I've never contributed any source to the FS community before. Thanks, Bob

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Re: [Freeswitch-dev] [Freeswitch-users] [Special Announcement] ClueCon Weekly Special Security Edition! Wed Oct 23rd @ 1PM Eastern
by Ken Rice 23 Oct '13

23 Oct '13

Again Thanks Cal! Today was great! Check out the links below for those that missed it! Hopefully this will get some people thinking about what not to do in the future and get some vendors to address things. Tl;dr version: Cal found a multi-vendor problem with auto-provisioning services... See the youtube link for the presentation. On 10/23/13 4:38 PM, "Cal Leeming [Simplicity Media Ltd]" <cal.leeming(a)simplicitymedialtd.co.uk> wrote: > For those that missed it, you can watch the whole thing here; > http://www.youtube.com/watch?v=raXkHi_uGF8 > > Slides here; > https://www.dropbox.com/s/hp5fj7e7o1mdnyt/Auto%20provisioning%20sucks.pptx > > Cal > > > On Tue, Oct 22, 2013 at 7:36 PM, Ken Rice <krice(a)freeswitch.org> wrote: >> ClueCon Weekly will present a Special Security Episode Tomorrow at 1PM >> Eastern Live! >> >> Cal Lemming, Will be making a 0day disclosure! >> >> Don¹t miss this exclusive presentation for the FreeSwitch community! >> >> This is one you will not want to miss! >> >> Join us on the FreeSWITCH Bridge for this Special Presentation! >> >> For conference call access information visit http://ss7.us/call888 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch

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freetdm + sangoma analog card A400，hangup problem
by openser 23 Oct '13

23 Oct '13

Hi all, recently, I use sangoma A400 analog for outgoing call, the flow is internal sip phone-> FXO->pstn. but, when the pstn side hangup first, the sip phone never be hangup, i search here and there, i also try use tone_detect app in dialplan , but ,still it can not detect busy tone, below is my dialplan: <action application="set" data="sip_h_X-accountcode=${accountcode}" /> 5 <action application="pre_answer"/> 7 <action application="tone_detect" data="busy 450 r 0 hangup 'normal_clearing' 4"/> 8 <action application="set" data="sip_h_X-Tag=" /> 9 <action application="set" data="call_direction=outbound" /> 10 <action application="set" data="hangup_after_bridge=true" /> 11 <action application="set" data="inherit_codec=true" /> 12 <action application="set" data="continue_on_fail=true" /> 13 <action application="bridge" data="freetdm/FXO/a/$1" /> my country busy tone frequency is 450, i never see detect busy tone in freeswitch log, what's more, when pstn side (modile ) hangup, the sip softphone will play DEFTONES continuly( filer rtp with wirshark, i cant play DEFTONES ) , but the call session never clear and no hangup signal got in freeswitch log. i check ftdm_io.c ,in function ftdm_channel_process_media, it will detect tone, but from freeswitch log, i can only see Dial tone be detect, below is code snap: if (ftdm_test_flag(ftdmchan, FTDM_CHANNEL_PROGRESS_DETECT) && !ftdm_channel_test_feature(ftdmchan, FTDM_CHANNEL_FEATURE_PROGRESS)) { uint32_t i; for (i = 1; i < FTDM_TONEMAP_INVALID; i++) { if (ftdmchan->span->tone_finder[i].tone_count) { if (ftdmchan->needed_tones[i] && teletone_multi_tone_detect(&ftdmchan->span->tone_finder[i], sln, (int)slen)) { if (++ftdmchan->detected_tones[i]) { ftdmchan->needed_tones[i] = 0; ftdmchan->detected_tones[0]++; } } } } } I know that , in asterisk , the zaptel have several option in deal with disconect surpervision, but it seems little in freetdm. any prompt is grateful! thanks.

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[Special Announcement] ClueCon Weekly Special Security Edition! Wed Oct 23rd @ 1PM Eastern
by Ken Rice 22 Oct '13

22 Oct '13

ClueCon Weekly will present a Special Security Episode Tomorrow at 1PM Eastern Live! Cal Lemming, Will be making a 0day disclosure! Don¹t miss this exclusive presentation for the FreeSwitch community! This is one you will not want to miss! Join us on the FreeSWITCH Bridge for this Special Presentation! For conference call access information visit http://ss7.us/call888 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://www.ss7.us/cluecon-gplus FB ClueCon : http://www.ss7.us/cluecon-fb G+ FreeSwitch : http://www.ss7.us/freeswitch-gplus FB FreeSWITCH : http://www.ss7.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch

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"Installation for Windows" wiki page is extra-protected
by Alex Forster 20 Oct '13

20 Oct '13

I'm trying to add more eloquently worded instructions on how to build the latest FreeSWITCH 1.2.14-stable to the "Installation for Windows" wiki page: http://pastebin.freeswitch.org/21536 This page appears to be more heavily protected than the general wiki. I can't edit this page, but I can edit any other page that I've found so far. Wiki page: http://wiki.freeswitch.org/wiki/Installation_for_Windows Read-only permissions message: http://i.imgur.com/oAOJP5H.png If this extra protection is by design, then ignore this email. Alex Forster

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Audio Quality Issues while playing back audio files
by Bharath Pareek 17 Oct '13

17 Oct '13

Hi, I have been trying to play mp3 and wav files using freeswitch but noticed that the audio quality of these files are degraded. After looking for a while on logs, I think the sampling rate of these files goes down to 8K as I see “Codec Activated L16@8000 1Channels 20ms” in the fs_cli log… How can I retain the audio quality (sampling rate) of these files ? -- Regards, Bharath Pareek

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question regard sip-reregistrations/renewals in sofia_reg.c
by Michel Brabants 16 Oct '13

16 Oct '13

Hello, I'm looking through sofia_reg.c about how reregistrations are being handled. More specifically, at lines 1599 - 1663. Link to file: http://fisheye.freeswitch.org/browse/freeswitch.git/src/mod/endpointss/mod_… I would think that during a renewal of a registration, when multiple registrations aren't allowed, you would also get an update of the current registration. What happens now seems to be a removal of the current registration in the database and an insertion of the "renewal". Lines 1605-1607 and 1638-1648. Does this mean that when the user is being looked up from another profile at the moment that the deletion just happened and the "reinsertion" not yet, that the user can't be found and a "not available" would be generated or am I wrong? Wouldn't an update be better in this case (or even considered a fix) or would this break code that I don't know about currently? I'm willing to submit the patch ...., but I need to know whether this would break anything. Kind regards, Michel

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FreeSWITCH news and Notes - FreeSWITCH 1.2.14 Released!
by Ken Rice 16 Oct '13

16 Oct '13

FreeSWITCHers! Today we are proud to announce FreeSWITCH 1.2.14! Available today via git, http://files.freeswitch.org/freeswitch-1.2.14.tar.bz2, and the deb and yum repos! This is a maintenance release to address several bugs that have been identified since the last release. Also dont forget ClueCon Weekly Conference Call! Every Wed at 1PM EST! For more information on how to join see: http://wiki.freeswitch.org/wiki/Weekly_Conference_Call_Calling_Instruct ions (What a coincidence! That¹s today!) -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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PORTAUDIO issue ?
by Thomas Chen 14 Oct '13

14 Oct '13

the board i am using FREESWITCH has CS5535/CS5536 Audio from shell.... i can play a WAV sound file without problem # aplay xxx.wav however, in fs_cli / portaudio, i have problem playing the sound file fs_cli > pa play xxx.wav (kernel complains about dma problem).... any idea/ suggestion where to look ? is portaudio doing something different from alsa utility (aplay) ??? thanks

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