- FreeSWITCH-users - FreeSWITCH Mailing Lists

PGP signatures for release tarballs
by David Matson 10 Nov '16

10 Nov '16

I'm maintaining some FreeSWITCH packages for Arch. If PGP signatures are available for source tarballs, the build system can use them to verify authenticity automatically. Would you be interested in producing a PGP .sig/.sign/.asc file alongside the release tarballs going forward? http://files.freeswitch.org/releases/freeswitch/ Thanks, David

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License for music and sounds
by David Matson 10 Nov '16

10 Nov '16

I'm maintaining some FreeSWITCH packages for Arch and have separate packages for music and sounds. What's the correct license to list for the audio tarballs available here: http://files.freeswitch.org/releases/sounds/ Is it MPL like the core code or something different? Thanks, David

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FreeSwich and SIP REFER
by Lợi Đặng 10 Nov '16

10 Nov '16

Hi list, I have this call scenario, it's just the simplest 2 sofia endpoints external profile, and a bridge application in dialplan Caller -> FS -> calee Then caller compose a REFER within current dialog, FS just handle that REFER, and contact the target sip uri indicated in Refer-To in REFER. Everything seems fine, since FS follow rfc3515 as a user agent, legs bridged Target transfer -> FS -> callee But I actually want my callee to receive and handle REFER message, any trick to do that? I'm thinking of creating new REFER within dialog to callee when FS receive a REFER from caller, but not sure if it's possible or how. My FS version is 1.6.12 rgds, Loi Dang Thanh Phone : 841224.735.448 Email : loi.dangthanh(a)gmail.com

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Adding users programatically
by Gonzalo Gasca Meza 09 Nov '16

09 Nov '16

Hi forum, I want to add sip users with mailbox (right now under /etc/freeswitch/directory/default as xml files) using FS API. I would like to store user information in database (using pgsql 9.5). Found these posts: http://voicebundle.com/how-to-add-sip-user-in-freeswitch https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/scripts/pe… http://voicebundle.com/creating-users-extensions-in-freeswitch-creating-sip… But these are scripts which manipulate XML files. Looking to store information in database and have freeswitch read this information. (I can take care of adding user, delete user, update user, handle duplicates, etc.). Similar like this post: http://telecommusings.blogspot.com/2009/10/using-modxmlcurl-in-freeswitch-h… Thank you

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bridge with esf_page_group ?
by Peter Hartmann 09 Nov '16

09 Nov '16

Is it possible to bridge a sip extension with esf_page_group? (the polycom multicast function) Whichever is first wins: (esf_paging_group or bridge) <extension name="rtp_multicast_page-PA_System"> <condition field="destination_number" expression="^4321$"> <action application="bridge" data="{sip_invite_domain=$${domain}}user/1234@${domain_name}"/> <action application="answer"/> <action application="set" data="esf_multicast_channel=50"/> <action application="set" data="esf_multicast_write_codec=G722"/> <action application="esf_page_group" data="224.0.1.116 5001"/> </condition> </extension I've tried loopback/1234 also to to avail. Also I've tried exec after bridge like so: <extension name="rtp_multicast_page-PA_System"> <condition field="destination_number" expression="^4321$"> <action application="answer"/> <action application="set" data="esf_multicast_channel=50"/> <action application="set" data="esf_multicast_write_codec=G722"/> <action application="set" data="exec_after_bridge_app=esf_page_group"/> <action application="set" data="exec_after_bridge_arg=224.0.1.116 5001"/> <action application="bridge" data="{sip_invite_domain=$${domain}}user/1234@${domain_name}"/> </condition> </extension> I tried 224.0.1.116 and 5001 each in their own argument statement. Any ideas how to achieve this? Thanks much, Peter Hartmann Hartmann Computer Consulting http://blog.hartmanncomputer.com (212)203-8870 Public Key ID 0x1acfea8b5d88088d If I can't explain it to you in plain language, that means I don't understand it.

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Default connection pool size for ODBC
by Muhammad Naseer Bhatti 09 Nov '16

09 Nov '16

Hi, what’s the default connection pool size for ODBC/Postgres is defined? -- Sent with Airmail

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mod_shout .mp4
by Anonim Stefan 09 Nov '16

09 Nov '16

Hi, I am trying to use mod_shout to play back the file [1]. I am using freeswitch version [2]. While doing this am getting error [3] and call eventually ends after ~ 10 secons with the error: "2016-11-08 13:31:45.946773 [ERR] mod_shout.c:437 Decoder Error! http://bc05.ajmn.me/665003303001/201605/19/665003303001_4906383897001_20160… " I am able to playback the stream using mod_vlc and vlc://http://LINK. Any ideas what might be causing this Decoder Error and how can I fix it? Thank you, Stefan [1] shout:// bc05.ajmn.me/665003303001/201605/19/665003303001_4906383897001_20160522-PG014904-V01-1-WIT-5818765.mp4 [2] FreeSWITCH version: 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit) [3] http://pastebin.com/YRsnVu70

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Unregister verto user
by Gregor Nanger 09 Nov '16

09 Nov '16

I am using verto with web clients without problems. It performs very well. But have one question. Is it possible to unregister verto user? Is there some kind of cli commad like verto xmlstatus? Best regards, Gregor

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bind_digit_action advice
by Gregor Nanger 08 Nov '16

08 Nov '16

Would need som advice. Just started to implementing bind_digit_action where caller can press 1 to transfer call during active call. Since I am using xml curl, I would like to minimize trip to server to get dialplan actions and make action inline not execute_extension. I would like to bridge call to user with extension 001, but having problem with syntax. Is this correct way? <action application="bind_digit_action" data=" inlinebridge , 1 , exec:bri d g e , user/001 "/>

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One way audio
by Agustí Ubalde 08 Nov '16

08 Nov '16

Hi all, I'm testing 2 video extensions trough FreeSWITCH but I only see the video at the caller browser. Called browser is receiving UDP packets but not show the video. Any sugestion? I'm using *FreesSWITCH 1.6.12* on *CentOS 7*. Thanks, *PRESENCE TECHNOLOGY* An *ENGHOUSE INTERACTIVE* Company *Agustí Ubalde Bellot* Chief Developer C/ Comte Urgell 240 3º-A Barcelona 08036 aubalde(a)presenceco.com Ph: +34 93 10 10 322 Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com <http://www.presenceco.com/>*

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