November 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH founder gets movie credit
by Ken Rice 08 Nov '14

08 Nov '14

New Post on freeswitch.org from anthm check it out at http://ift.tt/1ACmdcc FreeSWITCH founder gets movie credit I’m an extra in a movie but I’m not sure you ever see me. My son has a real part though as a kid named Sam and ends up in a sticky situation to say the least. Check out Available on iTunes and DVD! -Anthony Minessale

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Cannot Establish Audio Call First, Then Enable Video
by Jerry Richards 08 Nov '14

08 Nov '14

Using two Bria softphones, if I establish an audio call first, and then turn on video, the softphone sends a re-INVITE including the H264 codec, but Freeswitch does not forward the INVITE to the other endpoint. If I originate the call as a video call, then the video call is established successfully. I tried this with the latest Freeswitch as of yesterday. The only change I made to the default configuration was to add the H264 codec to <global_codec_prefs> and <outbound_codec_prefs> in vars.xml. Is there a configuration tag that would allow this? Or is it required that the video codec must be offered in the first INVITE for video to work (i.e. cannot be added later)? Regards, Jerry

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Newbie question: Set up to answer new phone number (please be gentle)
by T Fred Farmington 07 Nov '14

07 Nov '14

Since I am a total newbie to FreeSwitch please be patient and 'gentle' with me.... I have 'inherited' 2 FreeSwitch installations. One is working (version 1.0.7) and one (version 1.5.14) is a new test environment that is not working. I cannot get the new test environment to answer the inbound call and in the FreeSwitch log I find: Rejected by acl "domains" I have tried to search the FreeSwitch documentation for an answer and got lost in it. And I have tried to search the web for an answer, but did not find anything clearly spelled out. And, lastly, I tried to investigate the working installation, but could not find my answer. I have a dedicated SIP phone line which can call into the new test environment. And via FS_CLI I can see the call being received. However the connection is never established and in the log I find: Rejected by acl "domains" In both FreeSwitch environments I have a directory named: \Debug into which the previous individual installed all of the configuration parameters. Questions: 1. Where/How is FreeSwitch configured to recognize this directory and look for its parameters there? **** Maybe the non-working environment does not have that setting established and therefore is ignoring settings made there. 2. I have configured the new inbound IP address into the file: acl.conf.xml and have 'mirrored' the working environment in that regard. **** But the 'Rejected' message in the log suggests that somehow that is not being recognized. 3. Are there other files which I have missed in trying to set this up? Thanks

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Re: [Freeswitch-users] Registering FreeSWITCH with another server, multiples AORs
by Julio Cesar Esteves Cabezas 07 Nov '14

07 Nov '14

I have 100+ AORS to register with the main server. So, as far as I understand, I have to define 100+ gateways to the same main SIP server in FS, each one doing a single AOR registration. Is there any support for multiple AOR registrations per gateway ? Or any other more "compact" scheme ? Thanks. -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Thursday, November 06, 2014 11:52 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Registering FreeSWITCH with another server, multiples AORs The mod_sofia concept that makes outbound registrations is called a gateway. > On Nov 6, 2014, at 7:50 AM, Julio Cabezas <jcabezas(a)inovax.com.br> wrote: > > Hi, > How to make FreeSWITCH Sofia UA request several SIP registrations > (with different AORs) to an already deployed main SIP server ? > I need that for that server forward some calls to FS. > Thanks. _________________________________________________________________________ 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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FS as callback server pass through ring back tone form B leg to A Leg
by kthofer 07 Nov '14

07 Nov '14

Hi There we have FS as a Callback server right now we play our own RB tone $con->execute("set", "transfer_ringback \%(400,200,400,450);\%(400,2000,400,450)")right after a short welcome message. But that's not really how we like to have it. instead of our own RB tone we like to pass through the B-leg RB Tone to the A-leg. I know this is a question asked before and we tried to read through the forum. But all suggestions we found and tried did not work. WE can not make the Box passing through the RB tone from the b-leg Any suggestion or how to is highly appreciated. Hope to get some good answers -- With best regards Karl Theo Hofer M: +46 7030 22178 E: karl-theo_hofer(a)inteli-sim.com

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how to call asterisk extension from freeswitch using mod_event_socket.
by Aqs Younas 07 Nov '14

07 Nov '14

I want to call asterisk extension from freeswitch using ESL. Here is my python file. #!/usr/bin/env python import string import sys from ESL import * con = ESLconnection("localhost","8021","ClueCon") #are we connected? if con.connected: con.execute("bridge","sofia/external/555(a)192.168.1.44"); When a run this it does not generate call over asterisk extension. But when i do this using dialplan it works.Here is my default.xml <include> <context name="default"> <extension name="esl"> <condition> <action application="log" data="yes i did it"/> <action application="bridge" data="sofia/external/555(a)192.168.1.44 "/> </condition> </extension> </context> </include> *But i want to do this using mod_event_socket.* Your help would be much appreciated. Thanks

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[MOD Nibblebill] Postpaid Customer
by Massimo Varriale 07 Nov '14

07 Nov '14

Hi All, is possible to create a scenario in which I have a postpaid customer in which the credit can go below zero but no more than a maximum value allowed? The wiki talks about the feature of having a postpaid customer, but there are no example regarding this feature. But on prepaid there are many and many example, thank for this, great! BTW, currently I can create a prepaid customer and as soon as credit reach 0 the call is dropped correctly, but how can I implement a postpaid customer? Thank you so much Max

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 07 Nov '14

07 Nov '14

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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how to use mod_conference conference-flags and member-flags: video-bridge
by Daniel Kou 07 Nov '14

07 Nov '14

I can set conference-flags to audio-always|video-bridge , but I don't know how to join and control the video conference in this case. who can help me? thanks. Best regards Daniel

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Registering FreeSWITCH with another server, multiples AORs
by Julio Cabezas 06 Nov '14

06 Nov '14

Hi, How to make FreeSWITCH Sofia UA request several SIP registrations (with different AORs) to an already deployed main SIP server ? I need that for that server forward some calls to FS. Thanks.

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