July 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

script answers one session, creates another, need to hand off one to another lua without hanging up
by Trever L. Adams 21 Jul '15

21 Jul '15

Hello Everyone, I am stuck. I have a lua script that gets run from a bind digit setup. The only way to make it work was to have the bind digit call another lua script that contains: theret = api:executeString("uuid_dual_transfer " .. my_uuid .. " 'm:^:set:number=" .. call_id_num .. "^ set:name=" .. call_id_name .. "^lua:lua_script.lua/inline' " .. "ext/XML/context") The incoming call goes to the ext the lua script is a menu for the internal user. I have another script that answers some incoming calls. It then creates a new session and gives the internal user a menu. One option is to connect, another to voicemail. I have these working. The problem is that one option is to call the lua script above and to send the incoming call to the same ext. At one point I had it to where it would do one and then the other. I need a way to make them happen simultaneously. The lua script in the executeString above uses session, but the session is created as internal_session in the other script. i have tried bridging with execute_on_answer calling the script with the dual transfer above, but it executes AFTER hangup. Has anyone found a way to solve this problem? session:executeString doesn't seem to exist anymore for lua. Thank you, Trever

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FreeSWITCH Week in Review (Master Branch) July 11th-July 17th
by Ken Rice 20 Jul '15

20 Jul '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1JtgsMJ FreeSWITCH Week in Review (Master Branch) July 11th-July 17th Hello, again. This passed week in the FreeSWITCH master branch we had 43 commits. We had a number of cool new features this week including: added functionality for capturing screenshots from both legs to uuid_write_png, the addition of new multi-canvas and telepresence features in mod_conference, the addition of vmute member flag to mod_conference, and an API for removing an active ladspa effect on a channel. Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. New features that were added: FS-7769 [mod_conference] Add new multi-canvas and telepresence features FS-7847 [mod_conference] Add layers that do not match the aspect ration of conference by using the new hscale layer param for horizontal scale, and add zoom=true param to crop layer instead of letterbox, add grid-zoom layout group that demonstrates these layouts, and fix logo ratios and add borders too. FS-7813 [mod_conference] Add vmute member flag. FS-7846 [mod_dptools] Add eavesdrop_whisper_aleg=true and eavesdrop_whisper_bleg=true channel variables to allow you to start eavesdrop in whisper mode of specific call leg FS-7760 [mod_sofia] Revise channel fetch on nightmare transfer and add dial-prefix and absolute-dial-string to the nightmare xml FS-7829 [mod_opus] Add sprop-stereo fmtp param to specify if a sender is likely to send stereo or not so the receiver can safely downmix to mono to avoid wasting receiver resources FS-7830 [mod_opus] Added use-dtx param in config file (enables DTX on the encoder, announces in fmtp) FS-7824 [mod_png] Add functionality for capturing screenshots from both legs to uuid_write_png FS-7549 [mod_ladspa] Added an API for removing an active ladspa effect on a channel. For conformance reasons, the uuid_ladspa command now accepts ‘stop’ and ‘start’, while the previous functionality (without any verb) which will simply add ladspa remains intact. Improvements in build system, cross platform support, and packaging: FS-7845 [mod_conference] Break up mod_conference into multiple source files to improve build performance FS-7769 [mod_conference] Fixed a build issue FS-7820 Fix build system typo. Don’t assign the same variable twice. FS-7043 Fixed apr1 unresolved symbols in libfreeswitch.so.1.0.0 FS-7130 Make /run/freeswitch persistent in the Debian packages, so it will start under systemd The following bugs were squashed: FS-7849 [verto] Remove extra div breaking full screen in html FS-7832 [mod_opus] Fixes when comparing local and remote fmtp params FS-7731 [mod_xml_cdr] Fixed a curl default connection timeout FS-7844 Fix packet loss fraction when calculating loss average

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Mod_verto vs WebRTC(ws/wss)
by Murugan Pandian 20 Jul '15

20 Jul '15

Hi, Any one can help me what is difference between mod_verto and WebRTC(ws/wss) mod.

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mod_managed vs eventsocket
by Prashant Choudhary 20 Jul '15

20 Jul '15

Hello All, I have worked with LUA and freeswitch. Now I want to migrate to .Net API as I want to experiment with machine learning using sample calls. I am trying to control around some 2-300 calls directly through my API. This may require the call audio stream to be forwarded in real time to .NET I a few questions that might be very silly. 1. Which one is better to achieve this, mod_managed or ESL. 2. I was able to find some examples related to eventsocket, but same for mod_Managed was rare. Can you suggest me any doc(of-course except the wiki link) or any email chain which explains this or anything else. This will be really a big help for me. As currently I am stuck. And I still find the LUA way easier. Thanks

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Freeswitch WS+SIPJS+sip2sip
by Murugan Pandian 20 Jul '15

20 Jul '15

Hi, I try to register sip2sip.info account from browser using ws signalling,but i am getting SIP/2.0 403 Forbidden error in my console. The same sip2sip.info account work in all soft client

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Freeswitch+iptel
by Murugan Pandian 20 Jul '15

20 Jul '15

Hi, I need to use FS only media server and WS-SIP signalling server Media-server i disabled auth-call in internal.xml so i can make call from iptel account WS-SIP But when i try to use WS-SIP i can't able to register using iptel account getting 408 error(chrome browser) If i am doing wrong here pls help me

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Re: [Freeswitch-users] Is there a way to temporarily disable DTMF controls during an eavesdrop (or control an eavesdrop session with non-DTMF stimulus)?
by Steven Szeto 20 Jul '15

20 Jul '15

Michael Jerris asks: does it work if you do the set before the eavesdrop? That set is not running until after the eavesdrop ends. Answer: Yes, set:eavesdrop_enable_dtmf=false, does work if it is invoked before the eavesdrop. However there are two issues: * The "queue_dtmf" command is now ignored, which means the eavesdrop session starts as silent monitor instead of a whisper/coach session * The attempt to set eavesdrop_enable_dtmf=true does not seem to reopen the DTMF listening capabilities of the eavesdrop session. DTMF keystrokes 0,1,2,3 have no effect. The solution for me would be this: * Before the eavesdrop session is started, , set:eavesdrop_enable_dtmf=false * Introduce the ability to change the eavesdrop mode via the command line o E.g. eavesdrop_change_mode <uuid> [0,1,2,3] o Where [0,1,2,3] are the following eavesdrop modes respectively: silent monitor, coach the a-leg, coach the b-leg, barge-in conference From: Steven Szeto Sent: Friday, July 17, 2015 4:01 PM To: 'FreeSWITCH Users Help' <freeswitch-users(a)lists.freeswitch.org> Subject: Is there a way to temporarily disable DTMF controls during an eavesdrop (or control an eavesdrop session with non-DTMF stimulus)? I can successfully create a whisper/coach eavesdrop session with the following commands issued from fs_cli: originate sofia/internal/5401(a)10.47.41.109<mailto:sofia/internal/5401@10.47.41.109> &bridge(sofia/internal/5901(a)10.47.41.109<mailto:sofia/internal/5901@10.47.41.109>) whisper/coach: originate sofia/internal/5902(a)10.47.41.109<mailto:sofia/internal/5902@10.47.41.109> 'queue_dtmf:w2@500,eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861' inline I would like to temporarily disable DTMF so that the supervisor can not change the eavesdrop mode via DTMF, but this does not work: originate sofia/internal/5902(a)10.47.41.109<mailto:sofia/internal/5902@10.47.41.109> 'queue_dtmf:w2@500,eavesdrop:a28739d0-00f0-4a59-8c82-7a5a74ab6861, set:eavesdrop_enable_dtmf=false' inline Moreover, I have a GUI app that presents the supervisor with the ability to change the eavesdrop mode. When he presses the appropriate icon in the GUI, I would then like to reenable DTMF and change the eavesdrop mode on the session: Uuid_setvar a28739d0-00f0-4a59-8c82-7a5a74ab6861 eavesdrop_enable_dtmf true; Uuid_recv_dtmf a28739d0-00f0-4a59-8c82-7a5a74ab6861 w3@500 Is there another way to change the mode of an eavesdrop session? ________________________________ NOTE: This e-mail (including any attachments) is for the sole use of the intended recipient(s) and may contain information that is confidential and/or protected by legal privilege. Any unauthorized review, use, copy, disclosure or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please notify Mitel immediately and destroy all copies of this e-mail. Mitel does not accept any liability for breach of security, error or virus that may result from the transmission of this message.

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Serving Verto HTML from a different server
by Chad Phillips 20 Jul '15

20 Jul '15

I’ve gotten fairly comfortable with the setup offered in the Confluence wiki for the new Verto video features. I’m curious to know if I can expect any gotchas if I serve the Verto web stuff from a different server than the FreeSWITCH instance is running on. >From what I can tell, as long as I make sure to use the same server certs on both servers, it *should* work. Is there anything I’m not taking into consideration? Any better way to approach the problem besides certificate sharing across servers? Thanks, Chad

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I am using outbound to ivrd. My Call gets answered however why do I not get events with $conn->events("plain", "ALL");
by jaflong jaflong 20 Jul '15

20 Jul '15

<include> <extension name="demo"> <condition field="destination_number" expression="^(\d+)$"> <action application="set" data="ivr_path=/usr/local/freeswitch/scripts/ivr_test.php"/> <action application="socket" data="127.0.0.1:8004" async full/> </condition> </extension> </include> ivr_test.php -------------- #!/usr/bin/php <?php require_once('ESL.php'); $conn = new ESLconnection(0); $music = "/usr/local/freeswitch/sounds/demo.wav"; $conn->execute("answer"); $conn->execute("playback", $music); while ($conn->connected() == 1) { $conn->events("plain", "ALL"); $e = $conn->recvEvent(); if ($e) { $name = $e->getHeader('event-name'); //echo "Got event $name\n"; $out = fopen('php://stderr', 'w+'); fwrite($out, "Got event $name\n"); } else { //echo "Event empty\n"; $out = fopen('php://stderr', 'w+'); fwrite($out, "Event empty\n"); } } I am using outbound to ivrd. My Call gets answered however why do I not get events with $conn->events("plain", "ALL");

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shared line or other suggestion
by Jason Holden 20 Jul '15

20 Jul '15

I am currently trying to have the ability to place a call on hold and be able to pick the call up from a displayed list on a group of Polycom phones. I am currently using shared line but when the main phone places a call on hold the shared lines to not show the held call with the remote user CID but the primary phonesCID information. Does anyone have a recommendation that'll work better. I want to be able to see the remote CID of the caller who is on hold.

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