September 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

api_reporting_hook and session_in_hangup_hook issue
by David Ponzone 20 Sep '16

20 Sep '16

I am trying to run mod_curl after hangup, in order to sens some channel variables to a HTTP server: <extension name="from_SIP_account_to_US_TollFree"> <condition field="destination_number" expression="^00(1800\d+)$"> <action application="set" data="api_reporting_hook=curl http://server/script.php?orig=${sip_from_user}&dest=${destination_number}&c… headers get"/> <action application="set" data="session_in_hangup_hook=true"/> <action application="bridge" data="{sip_cid_type=none}sofia/gateway/trunk.out/+$1"/> </condition> </extension> When I hangup, only sip_from_user and destination_number are correct. answered_time contains 0 and hangup_time is empty Am I missing something ? Thank you David Ponzone Direction Technique email: david.ponzone(a)ipeva.fr <mailto:david.ponzone@ipeva.fr> tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr <blocked::http://www.ipeva.fr/> - www.ipeva-studio.com <blocked::http://www.ipeva-studio.com/> Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.

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Fax processing not successful - Result 5
by tarik＠pasifiktelekom.com.tr 20 Sep '16

20 Sep '16

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Debian 8 reboot now showing 2 free switch process
by Tim Meade 20 Sep '16

20 Sep '16

Occurred on 3 servers. After update and reboot FS starts two proceses. ps ax | grep free 1017 ? Ss 0:00 /usr/local/freeswitch/bin/freeswitch -ncwait -nonat -rp 1018 ? S<sl 0:01 /usr/local/freeswitch/bin/freeswitch -ncwait -nonat –rp Getting very eratic results. Fs_cli no longer connects. Any thoughts or ideas? FS code within last year. Thanks, Tim

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Digging into WebRTC for the first time
by Don Hawkins 20 Sep '16

20 Sep '16

I'm trying to get WebRTC going on our FS server for the first time. FS is not my problem though, WebRTC is. All I need is a one button script that dials into FS, at which point I will put the user in a conference, anyone have any example scripts that already do this? I've looked at all the free libraries and unfortunately java is not my thing so I've had no success. Thanks!

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Verto and ICE servers
by Oleg Stolyar 19 Sep '16

19 Sep '16

Hi guys, when I set iceServers to false, my calls stop working. Is there something in FS I need to set to make it work? My FS server has a public IP, so I don't think ICE should be required in this case. Is that true?

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Verto DTMF additional info
by Oleg Stolyar 19 Sep '16

19 Sep '16

Hi guys, Is there a way to pass additional info to FS with DTMF from Verto? With SIP I use DTMF INFO and add custom headers which are then available in FS. Can I do something like this in Verto?

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SIP Call from Videodevice
by Jürgen Wendler 19 Sep '16

19 Sep '16

Hello everyone, i’m trying to make a SIP call from an external device (e.g. SONY or POLYCOM) to the “default” 3500 conference room which has the video-mcu-profile attached. When the call is ringing it does not appear to gets really connected. When I use X-Lite or another SIP softphone everything is working so far. But with external hardware devices I am not able to join the conference room. Does anyone have experience with calls from hardware videoconf systems via SIP to freeswitch? Best Regards Jürgen

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mod_avmd fails to compile in FreeBSD 10.3
by Rajil Saraswat 19 Sep '16

19 Sep '16

Hello, The following error pops up while compiling: making all mod_avmd gmake[4]: Entering directory '/usr/home/fs/fs1.6/freeswitch/src/mod/applications/mod_avmd' CC mod_avmd_la-avmd_fast_acosf.lo avmd_fast_acosf.c:160:26: error: use of undeclared identifier 'MAP_POPULATE' MAP_SHARED | MAP_POPULATE, /* read-ahead on the file. Later accesses to the mapping ^ 1 error generated. gmake[4]: *** [Makefile:694: mod_avmd_la-avmd_fast_acosf.lo] Error 1 gmake[4]: Leaving directory '/usr/home/fs/fs1.6/freeswitch/src/mod/applications/mod_avmd' gmake[3]: *** [Makefile:666: mod_avmd-all] Error 1 gmake[3]: Leaving directory '/usr/home/fs/fs1.6/freeswitch/src/mod' gmake[2]: *** [Makefile:574: all-recursive] Error 1 gmake[2]: Leaving directory '/usr/home/fs/fs1.6/freeswitch/src' gmake[1]: *** [Makefile:3131: all-recursive] Error 1 gmake[1]: Leaving directory '/usr/home/fs/fs1.6/freeswitch' gmake: *** [Makefile:1410: all] Error 2 Any idea who to compile this? Thanks Rajil

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Quiet or Silent call
by Shawn Wheeler 19 Sep '16

19 Sep '16

I have a SIP devise which I set to call a pre-defined extension. I would like to have this device call the extension, however have the call silent on the device end with no ringing and muted on the receiving side. In other words, I want to execute the call from the device, the device speaker would not play a ringing tone, (forgive me if my wording it not correct). When the receiver picks up the call, the receiver is muted but has the ability to unmute their end of the call. Is this possible? If so is there recommendation on how to set this up? Thank you in advance. Shawn

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ssmtp and html email
by Rajil Saraswat 18 Sep '16

18 Sep '16

Hello, I use ssmtp for sending voicemail emails. Apparently, ssmtp can be used to send HTML emails as pointed out by http://stackoverflow.com/questions/34333643/using-ssmtp-to-send-html-e-mail…. How can i modify switch.conf.xml to incorporate this? At the moment it reads <param name="mailer-app" value="sendmail"/> <param name="mailer-app-args" value="-f my.emails.add(a)gmail.com"/> Thanks

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