October 2013 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] ClueCon Weekly News and Notes
by Ken Rice 29 Oct '13

29 Oct '13

Hey Guys, We had a great ClueCon Weekly Last week with Cal Leeming. This week we will be following up on last weeks Information with new information! Join us Wed at 1PM Eastern, 10AM Pacific via sip:888@conference.freeswitch.org or for other access see http://fs0.us/call888 And for all your social networking needs don¹t forget to join our circles on G+, our Group on FaceBook, and Follow us on Twitter! 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 Have a Great Week! Ken -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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[Freeswitch-dev] Improve mod_local_stream playlist
by Constantine 29 Oct '13

29 Oct '13

Now, as I see, playlist loc file can only contains absolute paths. I suggest one of this ways path can be relative (e.g. relative to ${sounds_dir}) playlist can be requested using mod_curl local_stream conf could contain playlist tag -- With Best Regards, Constantine

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[Freeswitch-dev] How to use "say" command for a conference
by Prevail Magid 28 Oct '13

28 Oct '13

Hi All, I try to play in conference some message by using freeswitch console api. I tried enter next commands : "*conference my_conf_name say my_custom_text*", "*conference my_conf_name say 'en ip_address pronounced 12.34.56.78*" and get response *(say) Error!* *say <text>* I use freeswitch version 1.2.1. Modules  <load module="mod_flite"/>  <load module="mod_say_en"/> Note: When I use 'say <text>' in dialplan all work fine.

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[Freeswitch-dev] 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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[Freeswitch-dev] 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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[Freeswitch-dev] 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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[Freeswitch-dev] [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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[Freeswitch-dev] "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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[Freeswitch-dev] 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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