September 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

what can I have in ${} construction
by covici＠ccs.covici.com 11 Nov '14

11 Nov '14

Hi. I was experimenting with ${variable} and was wondering what could be in there before the }? I can do a substring if I say variable:2, but I wonder what else you can do? The bash manual has all kinds of things you can have, but how much of this will fs do? Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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XML DialPlan help
by Sean Devoy 04 Nov '14

04 Nov '14

I swear I have read this, but now that I need it I can't find it. My user wants to dial international. I want to require them to do SOME extra step other than dialing 0113010... to call the Netherlands. Oddly enough, it is cheaper than US termination - go figure. Ideally, I would like them dial some extension like 777, hear a new dialtone and then accept the international number. I might even like to play a message that says "All charges for International are their responsibility and not part of their current package" before the dialtone. Ideas? Thanks again, Sean

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Embedded Freeswitch
by Guillermo Ruiz Camauer 04 Nov '14

04 Nov '14

Can anyone point me to any examples / source code of embedde Freeswitch? I have looked at the documentation, and the different books, but can find now examples of this. Guillermo -- Guillermo Ruiz Camauer

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Logging commands
by Oleg Stolyar 28 Oct '14

28 Oct '14

Guys, is there a way to log cli commands that are being executed? More specifically I use the txtapi command from the xml_rpc module and need to be able to see when FreeSWITCH received and executed each command with argument. Is it doable already or would it require a code change? I already have this in my logfile.conf.xml <map name="all" value="debug,info,notice,warning,err,crit,alert"/>

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Windows server + freeswitch
by Guilherme Dantas 27 Oct '14

27 Oct '14

Hello, I am new at this mail list, so I don`t know how to check if this question was asked before (if yes, I am sorry) Which of these two Windows Server versions would be better to use with the Freeswitch? - Windows server 2008 R2 or - Windows server 2012 Thanks in advance, <http://www.inovax.com.br/> Description: Description: www.inovax.com.br Guilherme Vieira Dantas - Estagiário <mailto:guilherme.dantas@inovax.com.br> guilherme.dantas(a)inovax.com.br Skype: inovax5554 _____ Rua Paulo Emídio Barbosa, 485, Quadra 1, Centro de Excelência TIC Ilha da Cidade Universitária Rio de Janeiro RJ - CEP 21941-907

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LDAP
by Nick Zaitsev 08 Oct '14

08 Oct '14

Good day to you. Could you advise me how i can implement auth users via LDAP in the simplest way? Do i have to write my own application for mod_xml_curl(maybe there is such application somewhere)? Or i can use the mod for freeswitch and it will work like a charm.(i use both sip and skinny users) Thank you for your time, Best Regards Nick Zaitsev

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create dummy channel to record conference
by Alexandre Mouillé 07 Oct '14

07 Oct '14

Hi, I hope I'm not posting this in the wrong mailing list. I'm working on a custom conference that uses the conference module. I need a record function that keeps recording (or at list appends the same wav file) when no one is in a given conference. The solution that was used in the Asterisk program I'm porting was to create a dummy channel that goes in the conference and records what it hears in a file. - Can I create a dummy channel with an "originate /loopback" that would go into a conf and record what it hears (I'm not sure what loopback really is) - Otherwise, can I use originate from the FS box to call itself, one side connects to the conference and the other records. (I'm not sure how to do that, and how I could transmit variables (to tell which file to record to) to the called program) Thanks

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FreeSwith does not setup secure call with leg B.
by Chandrakant Marathe 05 Oct '14

05 Oct '14

Hi All, I have setup FreeSwith PBX. I am facing an issue of not having end to end secure call. Caller sends INVITE request with SIPS in request URI and RTP/SAVP in SDP. But when FreeSwith forwards the request to caller, it is not using RTP/SAVP in SDP. I have followed the steps mentioned in WIKI ( https://wiki.freeswitch.org/wiki/SIP_TLS#Configuration) for TLS configuration and marked "exports sip_secure_media" true in "conf/dialplan/default.xml" file (as per https://wiki.freeswitch.org/wiki/SRTP) But still FreeSwitch does not use RTP/SAVP for leg-B. Also, I am bit confused with following condition in "default.xml" when wiki page suggests that late coded negotiation must NOT to be turned on.  <condition field="${endpoint_disposition}" expression="^(DELAYED NEGOTIATION)"/> <condition field="${switch_r_sdp}" expression="(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)" break="never"> <action application="set" data="sip_secure_media=true"/>  <action application="export" data="sip_secure_media=true"/> </condition> By commenting/un-commenting "sip_secure_media=true" above, it did not worked. Any help with this would be greatly appreciated. -- Thanks CWM

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continue_on_fail vs fail_on_single_reject
by Andre 03 Oct '14

03 Oct '14

Hi, Can someone tell me when I should be using fail_on_single_reject over continue_on_fail? Let's say I have 2 bridges and if the call is busy I just want to hang up the code below should be correct? <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE"/> <action application="bridge" data="sofia/test-int/3910(a)10.3.3.104"/> <action application="bridge" data="sofia/test-int/3920(a)10.3.3.104"/> Now let's say the code looks like this Would I use <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE"/> or <action application="set" data="fail_on_single_reject=USER_BUSY"/> <action application="bridge" data="{fail_on_single_reject=USER_BUSY}sofia/test-int/3920(a)10.3.3.104| sofia/test-int/3920(a)10.3.3.104"/> Or <action application="bridge" data="{ continue_on_fail=NORMAL_TEMPORARY_FAILURE }sofia/test-int/3920(a)10.3.3.104| sofia/test-int/3920(a)10.3.3.104"/> Understanding that NORMAL_TEMPORARY_FAILURE it not totally like USER_Busy but just trying to understand when to use what. Thanks Andre

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Video conferencing with Bria 4 for Windows
by Stanislav Sinyagin 02 Oct '14

02 Oct '14

I found out that Bria 4 for Windows ignores the SIP INFO with picture_fast_update, answering with 500 error. As a result, in a video conference, when active floor holder switches to a Bria user, everyone sees garbage. Tested with both VP8 and H264 codecs. Bria for Android accepts the INFO and refreshes the picture properly. I also opened a topic on their support forum: https://support.counterpath.com/responses/bria-4-ignores-sip-info-with-pict… Any tips, ideas, suggestions to make the conferencing work under Windows? thanks, stanislav

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