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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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mod_dptools/mod_spandsp and tone detection
by Sasha Pachev 06 Nov '14

06 Nov '14

I have discovered an issue when using mod_dptools and mod_spandsp together. Both define app "stop_tone_detect". mod_dptools defines "tone_detect" to start tone detection, mod_spandsp defines "start_tone_detect". Both modules load successfully, and the conflict is resolved by awarding whoever loads the last the entry for stop_tone_detect. This creates problems - for example, if mod_spandsp wins, and the dialplan uses tone_detect/stop_tone_detect, then mod_dptools app will be called for starting, and mod_spandsp will be called for stopping, which in effect means it is a no-op. Is it considered an error to load both modules in the same configuration? What is the recommended solution/workaround if you want to use some of the functionality of mod_dptools and some of the functionality of mod_spandsp?

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GIT down?
by Daniel Ivanov 06 Nov '14

06 Nov '14

I am getting : git clone -b v1.2.stable git://git.freeswitch.org/freeswitch.git Initialized empty Git repository in /usr/local/src/freeswitch/.git/ git.freeswitch.org[0: 209.105.235.6]: errno=Connection timed out fatal: unable to connect a socket (Connection timed out) on 2 separate machines on different networks. Is it possible the git repo is down?

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How to originate call over another server using ESL
by Aqs Younas 06 Nov '14

06 Nov '14

Hi, I want freeswitch to send a call to asterisk's extension running over some different ip using ESL. But does not know how to do so.Can anybody help? I have successfully installed ESL for python.

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