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Problems with wait_for_silence API call
by Matthew 13 Jun '16

13 Jun '16

Hello All, I've having a problem with the wait_for_silence API call in lua. When I attempt to orginate a call and then check for silence on the line, I'm getting unpredictable results. Sometimes it seems to work, but more times than not it times out ( switch_ivr_wait_for_silence: TIMEOUT 0 ). To test this, I answer the call, speak a short greeting and then mute the handset. In other words, I know the script should be detecting silence on the other end of the call. Here is a good portion of the test script I'm using ... local cid_name = argv[1] local cid_number = argv[2] local gw_string = argv[3] local queue_ext = argv[4] dialout_string = "{origination_caller_id_name='" .. cid_name .. "'," .. "origination_caller_id_number=" .. cid_number .. "," .. "ignore_early_media=true}" .. gw_string dialout_session = freeswitch.Session( dialout_string ) while dialout_session:ready() and not dialout_session:answered() do freeswitch.msleep( 250 ) end if dialout_session:ready() and dialout_session:answered() then dialout_session:consoleLog( "INFO", "dialout: connected to " .. gw_string ) ) dialout_session:execute( "voice_start" ) dialout_session:sleep( 2000 ) dialout_session:execute( "wait_for_silence", "300 30 15 15000" ) dialout_session:execute( "voice_stop" ) dialout_session:consoleLog( "INFO", "dialout: amd_detect = " .. amd_detect ) end And here is an example of how I would invoke the script ... luarun dialout.lua TEST 1234567890 sofia/gateway/twilio/+1234567890 7000 Any ideas why the wait_for_silence API call would be misbehaving? Any suggestions on how I could debug this problem further? Thanks, -Matthew

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Channel/session variables availability on execute_on_sip_reinvite
by Florent Krieg 13 Jun '16

13 Jun '16

Hello everybody, My use case is pretty simple: I would like to react on SIP Re-INVITE without sdp (empty body/content-length 0) so I have put an execute_on_sip_reinvite instruction in my bridge string. I have tried both to call the 'info' app and to call a custom shell system script, passing the ${uuid} var and calling fs_cli uuid_dump $uuid, but unfortunately it seems that when the SIP message reaches the application it has been processed by the Sofia stack and I can't detect if there is an SDP or not (using sdp variables, or any other actually). Sometimes it gives me 0.0.0.0 as the remote_media_ip, sometimes it just builds an SDP based on the previous one. I have also tried to modify the re-INVITE by adding custom headers or putting information in any mandatory SIP header (like, say, Via), but these updates doesn't seem to be available via any channel variable. Is there any FS configuration parameter I could change to be able to access this information? Is there any other way I could do what I want? Thanks in advance if you have any clue :) Florent

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Need help on call bridge !!
by turja 13 Jun '16

13 Jun '16

Hello i am new user of Freeswitch. i am trying to write a dial plan that when i call to 1234(callee) first call bridge to 1233(another number) after 1233 hangup the call bridge 1234(callee) ....but can not ...after hangup from 1233(another number)the call hangs up not bridging with 1234(callee)..... how to do that in freeswitch. Thanks for help ! Turja -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Need-help-on-call-bridge-tp75… Sent from the freeswitch-users mailing list archive at Nabble.com.

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add modules when installing from the latest branch
by Matt Broad 13 Jun '16

13 Jun '16

Hi, this may be a really stupid question but here goes: I have upgraded a server to debian 8 and have installed freeswitch using the latest branch https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie#Debian… : My question is how do I activate other modules? In previous versions I would have amended the moudules conf file and re-compile. I can't find the modules file in this install thanks Matt

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ReInvte in bypass/proxy mode | a > a+v
by Mirko Brankovic 13 Jun '16

13 Jun '16

Hi, Can someone help me figure out the problem i am hitting. Im trying to upgrade call from audio to audio + video (opus 48k and vp8- but this one is only in passtrough mode). So set up is browser (sip.js/node.js) >> kamailio wss >> freswitch (bypass mode bridge without answer) >> kamailio >> browser (node.js) Problem is that freeswitch receives reInvite on existing audio call with new sdp containing also video codecs, but directly from client so no proxy (kamailio) in path, and exiting invite towards kamailio for outgoing leg is missing video in a=group BUNDLE, so only audio and a=sendrcv is also being stripped (that is the difference to 1 on 1 video call). I understood that bypass mode shouldn't change/touch sdp. Also passtrough codec shouldn't be the problem i guess since both clients can decode it. Any help is appreciated. Thanks, Mirko

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SHA-2 Implementation
by Abdul Hakeem 12 Jun '16

12 Jun '16

Hello, Any plans of moving FS from SHA-1 to SHA-2 crypto hash support ? I understand SHA-1 will be deprecated in Chrome by December, this will cause all calls to Verto to fail. Cheers, AH

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Authentication of SIP phones against LDAP directory?
by Kevin Long 12 Jun '16

12 Jun '16

Hello, Can Freeswitch authenticate SIP phone logins from an LDAP directory ? Hoping to integrate this so my SIP users can use the same password/username they use for all other applications. Regards, Kevin Long

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Auto record by call direction
by Ashwin Rath 12 Jun '16

12 Jun '16

Hello To record a call (inbound or outbound) i set the following variable for a user in the directory folder <variable name="user_record" value="enabled"/> However i would like to record inbound calls only. Is there some way to selectively auto record inbound calls only ?

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Re: [Freeswitch-users] mod_gsmopen: are USB dongles the only option?
by Giovanni Maruzzelli 12 Jun '16

12 Jun '16

Ciao Stanislav, the whole idea of gsmopen is to use cheap and ready available radio interfaces, that maybe you can get off ebay or surplus. Also, they are undistinguishable from "normally used" dongles, and that is a plus in a lot of countries. I don't know about pci modems that exposes an audio interface to the operating system. Some soldering and demuxing will be probably needed. But at that point, I would go with Samgoma or Khomp cards... Also because for doing it well, a lot of engineering details are involved, from shielding to low pass filtering, to magnetic interferences, etc Anyway, I like those things too! :) -giovanni sent from mobile cell: +39 347 266 56 18 Giovanni Maruzzelli OpenTelecom.IT Il 05/Giu/2016 16:06, "Stanislav Sinyagin" <ssinyagin(a)gmail.com> ha scritto: does anyone know a GSM/UMTS/LTE modem that has mini-PCIe interface and is able to deliver voice signal that can be passed to mod_gsmopen? The only one that has any kind of voice signal that I found, is Huawei MU609 and the like. But they deliver PCM voice as a raw signal on mini-PCIe pins which are marked as "reserved" in the interface specifications, so normally the boards don't have any wiring for those pins. I just don't like the idea of an USB dongle, and there are many hardware platforms with mini-PCIe interfaces. They work perfectly for mobile data, but I couldn't find any option for voice. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Dynamic Group Call
by Mititelu Stefan 12 Jun '16

12 Jun '16

Hi, I'm a Freeswitch newbee and I'm trying to set up dynamic group calls. Freeswitch will receive a header with "Group-Users:user1,user2.." and what I want to do is to "transfer" to a dinamycally created context that will look like: <action application="conference_set_auto_outcall" data="user1"/> <action application="conference_set_auto_outcall" data="user2"/> Is this possible? Can you give me some references? >From what I see in some examples one can transfer only to existent XML context: <action application="transfer" data="1105 XML default"/> Thanks, Stefan

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