- FreeSWITCH-users - FreeSWITCH Mailing Lists

Does xml_curl replace the entire directory or just that of the user in question
by Oivvio Polite 08 May '16

08 May '16

xml_curl passes the username for the user trying to authenticate as one of several arguments in an HTTP header. But when the answer comes back from the webservice does FS use that to replace the entire existing user directory or does it just replace the entry of the user in question? Oivvio

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Re: [Freeswitch-users] Calling multiple endpoints on an incoming call
by Stanislav Sinyagin 08 May '16

08 May '16

group_call returns a combined dial string that makes a call to its members. You can produce such a string yourself, and the end result will be the same. So, if you have an external application that controls the dialplan, it can create such strings directly, without using the groups in the directory. On May 5, 2016 21:04, "Richard Mace" <richard.mace(a)gmail.com> wrote: Hi, I am just reading through the FreeSWITCH 1.6 cookbook and I'm a little confused on calling multiple endpoints. If I have an incoming call and I want to ring 3 extensions, is it better to bridge to a group or ring the multiple end points by specifying them individually? Which situation would be a fit for each one? Thanks Richard Mace RoCKSoftware _________________________________________________________________________ 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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JSON Web Tokens
by Oivvio Polite 06 May '16

06 May '16

Can FreeSwitch handle JSON Web Tokens natively or be made to handle JWT through one of the available scripting languages? Oivvio

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FreeSwitch Dialplan and javascript
by Nduwayezu, Joselyne 06 May '16

06 May '16

Hello, I,m invoking a script (javascript) in the dialplan, but when i call my DID number, it rings busy and hsow the error: Responding to INVITE with 480. I've tried many time to fix it, but i still hane no results. I've edit an extention in /public directory, no resluts; add an extension in public.xml in the dialplan directory, still have no results. Can you please tell me the steps i have to follow in order to use my script in the dialplan? NDUWAYEZU Joselyne -- Haute École de Bruxelles

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Passing stuff from javascript to mod_xml_curl and then to a authentication web service
by Oivvio Polite 06 May '16

06 May '16

Today I've been trying to grok mod_xml_curl. I've set up a webservice that xml_curl will call to get the user directory. I've figured out from experimentation that my endpoint gets called whenever a sip user tries to register or in my case when verto is initialized in the browser. What I haven't figured out is how to pass stuff from the browser to mod_xml_curl and then on to the webservice. In my case I want to pass a JSON Web Token. I can pass it as the passwd argument in the verto initialization. vertoHandle = new jQuery.verto({ login: 'user@mydomain', passwd: 'xxxxxx', socketUrl: 'wss://mydomain:8082', deviceParams: { useMic: true, useSpeak: true }, iceServers: true }, vertoCallbacks); But how then do I pick that up in in my xml_curl conf and pass it on to the webservice? <configuration name="xml_curl.conf" description="cURL XML Gateway"> <bindings> <binding name="directory"> <param name="gateway-url" value="https://127.0.0.1/my_jwt_checking_webservice" bindings="directory"/> </binding> </bindings> </configuration> The webservice will be running on the same box as FS so passing the token to it as a URL paramater or as a HTTP header would be equally fine by me, but right now I can't even figure out how to access it. Regards Oivvio

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RFC 5626 SIP Outbound
by Colin Morelli 06 May '16

06 May '16

Hey all, Is there anything special that needs to be done to get Freeswitch/Sofia to support RFC 5626 for inbound registrations? I have a PJSIP client sending "Supports: outbound" with a reg-id and sip instance, but FS doesn't respond with Require: outbound and so PJSIP appears to think SIP outbound isn't enabled. I see the Sofia param rfc-5626, but that only seems to affect outbound registrations to upstream SIP providers that FS is registering with when acting as a UAC. Best, Colin

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bad udp cksum
by Nduwayezu, Joselyne 06 May '16

06 May '16

What means bad udp cksum error when in freeswitch. I can not get the incoming call when i invoke a script (janascript) in the dialplan. i get errors like: bad cksum adn then : 480 Temporarily unvailable Responding to INVITE with 480 Any ideas? NDUWAYEZU Joselyne -- Haute École de Bruxelles

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Video with Windows Server
by Jack Loranger 06 May '16

06 May '16

Is it possible to serve video from a windows server? Thanks, Jack

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High load kernel
by Agustí Ubalde 06 May '16

06 May '16

Hi all, We are having problems with the CPU load of Freeswitch. Specifically, there is a CPU having a charge of 70% of CPU system, always in the same CPU. Any ideas on the possible cause of this system CPU consumption? Thanks, *PRESENCE TECHNOLOGY* *Agustí Ubalde Bellot* Chief Developer C/ Comte Urgell 240 3A Barcelona 08036 aubalde(a)presenceco.com Ph: +34 93 10 10 300 Fx: +34 93 10 10 333 *www.presenceco.com* <http://www.presenceco.com/> *Follow us on:* *[image: tw]* <http://twitter.com/presencetech>*[image: yt]* <http://www.youtube.com/PresenceTechnology>*[image: in]* <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]* <http://www.slideshare.net/PresenceTech>*[image: fb]* <http://www.facebook.com/presencetech> For additional information, please visit our website *www.presenceco.com* <http://www.presenceco.com/> -- *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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Pulse / portaudio / sharing
by Rick Jarvis 06 May '16

06 May '16

I’d like to monitor the audio levels on an audio interface that’s being used by mod_portaudio I’m guessing, but I don’t know, that the best way of doing this would be to use Pulse Audio, either directly or indirectly. I don’t know a huge deal about Pulse Audio though, and while it seems that there’s a mod_pulseaudio available, I can’t find much information on it, and there seem to be quite old unanswered questions about Pulse in relation to FreeSWITCH, on this list. Can anyone give me any suggestions on which way I might head?

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