August 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Inter-domain connection
by Alois Komenda 05 Aug '08

05 Aug '08

Hello, how do I realize an inter-domain connection with FreeSWITCH? I have two instances of FS. One in domain A and one in domain B. How can users in A call users in B (and vice versa)? Thanks in advance! Alois Komenda Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK

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recording in mono
by Ilan Perez 05 Aug '08

05 Aug '08

What setting do I change to change the record_session code to record in mono I have edited this line and it doesn't seem to do much. from <action application="set" data="RECORD_STEREO=true"/> To <action application="set" data="RECORD_STEREO=false"/> Anyone. Ilan Perez

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Re: [Freeswitch-users] inbound calls
by Roberto Hernandez 05 Aug '08

05 Aug '08

Figured it out. Thanks, Roberto _____ From: Roberto Hernandez [mailto:rh@bluevisor.com] Sent: Monday, August 04, 2008 11:01 AM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: inbound calls I've configured one X-Lite softphone (id=1000) and made my first few PSTN outbound calls with no problem. Inbound is a bit of a mystery. How do I configure PSTN inbound calls to terminated on my softphone? I found the following archived email, would someone break it down so new users can understand? Thanks in advance. In public.xml, you want something like this: <extension name="test_did"> <condition field="destination_number" expression="^(4153084258)$"> <action application="transfer" data="$1 XML default"/> </condition> </extension> and then in default.xml have something like this in the default context: <extension name="Local_Extension"> <condition field="destination_number" expression="^(4153084258)$" continue="on-true"> <action application="set" data="dialed_ext=$1"/> </condition> <condition field="destination_number" expression="^${caller_id_number}$"> <action application="set" data="voicemail_authorized=${sip_authorized}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default $${domain} ${dialed_ ext}"/> <anti-action application="ring_ready"/> <anti-action application="set" data="call_timeout=10"/> <anti-action application="set" data="hangup_after_bridge=true"/> <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="bridge" data="USER/${dialed_ext}@$${domain}"/> <anti-action application="answer"/> <anti-action application="sleep" data="1000"/> <anti-action application="voicemail" data="default $${domain} ${dialed_e xt}"/> </condition> </extension> Thanks, Roberto

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logging
by Simon Shaw 04 Aug '08

04 Aug '08

Is it possible to declare which modules and which level of logs you want to be printed to the console? Also, is it possible to redirect the logs to an IP address? Thanks, Simon

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inbound calls
by Roberto Hernandez 04 Aug '08

04 Aug '08

I've configured one X-Lite softphone (id=1000) and made my first few PSTN outbound calls with no problem. Inbound is a bit of a mystery. How do I configure PSTN inbound calls to terminated on my softphone? I found the following archived email, would someone break it down so new users can understand? Thanks in advance. In public.xml, you want something like this: <extension name="test_did"> <condition field="destination_number" expression="^(4153084258)$"> <action application="transfer" data="$1 XML default"/> </condition> </extension> and then in default.xml have something like this in the default context: <extension name="Local_Extension"> <condition field="destination_number" expression="^(4153084258)$" continue="on-true"> <action application="set" data="dialed_ext=$1"/> </condition> <condition field="destination_number" expression="^${caller_id_number}$"> <action application="set" data="voicemail_authorized=${sip_authorized}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default $${domain} ${dialed_ ext}"/> <anti-action application="ring_ready"/> <anti-action application="set" data="call_timeout=10"/> <anti-action application="set" data="hangup_after_bridge=true"/> <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="bridge" data="USER/${dialed_ext}@$${domain}"/> <anti-action application="answer"/> <anti-action application="sleep" data="1000"/> <anti-action application="voicemail" data="default $${domain} ${dialed_e xt}"/> </condition> </extension> Thanks, Roberto

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ANNOUNCEMENT: Cluecon this week, no official docs conference call
by Michael Collins 04 Aug '08

04 Aug '08

FYI, Since many of the regulars will be at Cluecon this week we won't be on the documentation conference call. However, if those who aren't going to Cluecon are able to meet at the regular time then by all means do so. As a suggestion I would say to check out the results of last Wednesday's "mod of the week": http://wiki.freeswitch.org/wiki/Mod_cdr_csv Other suggestions: continue documenting whatever you can, gather together more documentation suggestions. We'll all reconvene next week. Thanks! -MC

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Re: [Freeswitch-users] Complete seperation of VOIP app from FS
by Erol Akarsu 04 Aug '08

04 Aug '08

David, How a remote process, connected to FS through mod_socket, can get CDR record of last call it made? Thanks ----- Original Message ---- From: David Knell <dave(a)3c.co.uk> To: freeswitch-users(a)lists.freeswitch.org Sent: Friday, August 1, 2008 12:19:49 AM Subject: Re: [Freeswitch-users] Complete seperation of VOIP app from FS Hi Erol, A dialplan extension like this: <extension name="callthrough"> <condition field="destination_number" expression="^0"> <action application="socket" data="127.0.0.1:6100 async full" /> </condition> </extension> results in all calls to numbers starting with 0 being handled by a server on localhost listening on port 6100. That server has full control over the call. Change the IP address to run the server on a different host. Cheers -- Dave > Still searching event_socket module. > If we set up dialplan, all received calls will routed based on dialplan rules. > But at the same time, if there is remote process connected to FS through socket_module, dialplan or remotes process will have priority to control FS behavior. Can remote process interfere dialplan rules.? > > Is it possible to define a dialplan so that one extension will be routed to a remote process and other extension will be routed to different process? > > I am thinking to abstract out IVR or any VOIP application from FS box completely and to run application on different host or process. > > Regards > > Erol Akarsu > > > > > > > _______________________________________________ > 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 > > > -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk _______________________________________________ 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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How to setup TLS
by Peter P GMX 04 Aug '08

04 Aug '08

Hello, has anybody managed to setup TLS? When I change tls to "true" in internal.xml, then freeswitch doens't listen on any ports (5060 5061). I use freeswitch 1.0.0-0ubuntu1~ppa4 Is there any tutorial available (could not find it while googling)? I would like to set it up with Snom phones (Snom320) Best regards Peter

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FW: unit testing
by Simon Shaw 04 Aug '08

04 Aug '08

I tried sending this question to the development list but the message fails to appear on the list and I receive no errors. Thought you guys might have an answer to this question anyway. Thanks, Simon From: Simon Shaw Sent: Monday, August 04, 2008 1:02 PM To: 'freeswitch-dev(a)lists.freeswitch.org' Subject: unit testing Does freeswitch have a unit testing policy? If so what is the code coverage like? Thanks, Simon

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FS with Openser
by ram 04 Aug '08

04 Aug '08

Hi we are looking to integrate freeswitch and Openser Openser as a frontend and Freeswitch to handle Media / and NAT and CDR any one can point me some documents for integrating this ram

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