October 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Digest Authentication issue
by Masood Ahmed 20 Oct '11

20 Oct '11

Dear , I had successfully installed and configured radiusclient. I do not know that what next lines should i do to get users registered on my freeswitch using digest authentication with my radius server, Like I have my radsius server and billing system. Now what lines should i add in sofit conf or in dialplans that every time client hits server request should be sent to radius server and what lines should i add to register them if i get proper return code and how to do that. Your Help is needed

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Freeswitch handling 300 multiple choice redirect
by Darren DeLitizia 19 Oct '11

19 Oct '11

All, I am having a hell of a time getting Freeswitch to handle multiple choice 300 redirects. I've followed directions on --> http://wiki.freeswitch.org/wiki/SBC_FreeSWITCH_Configuration_Example_2 but it does not work. It never gets past the first contact. I've done Googled my arse off and still nothing. Can anyone help? Currently my config is exactly what you see on that link above. I am using F.S. as topology hiding entrance point and Opensips as LCR redirect. The call flow is Customer --> F.S. --> Vendors from redirect | | Opensips (returns 300 multiple) Thanks DD

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esl_event_get_header error
by adam harris 19 Oct '11

19 Oct '11

I am trying to check if event received was CHANNEL_HANUP I am doing somthing like this const char *last_event = esl_event_get_header(call_info->global_handle->last_event,"Event-Name"); if (strcmp(last_event,"CHANNEL_HANGUP")) { exit(0); } However an error occurs src/esl_event.c:244: esl_event_get_header_ptr: Assertion `event' failed.

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Proxying video stream, recording audo stream
by Dmitry Matveev 19 Oct '11

19 Oct '11

Hello, I have a VoIP client software with a video codec unsupported by FreeSWITCH. To make video calls possible, I am currently using 'bypass_media' setting. But I would like to record all audio calls with FreeSWITCH. Is it possible to configure FreeSWITCH to 1. Proxy and process (record) audio RTP streams, and 2. Proxy (but not process) video RTP streams? Thanks in advance, Dmitry

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FreeSWITCH Conference Call Today
by Michael Collins 19 Oct '11

19 Oct '11

Hello all, Our agenda today is light: http://wiki.freeswitch.org/wiki/FS_weekly_2011_10_19 We're going to have a community scrum where we ask each other questions, etc. I have a few minor updates for you all as well. Talk to you soon. -Michael

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Re: [Freeswitch-users] Kamailio Dispatcher fails in front of FreeSwitch, help needed.
by Henrik Aagaard Sørensen 19 Oct '11

19 Oct '11

My dispatcher.list only contains: 1 pbx.my-domain.com:5060 On Oct 18, 2011 11:04 AM, "vip killa" <vipkilla(a)gmail.com> wrote:

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Re: [Freeswitch-users] Different SAS with ZRTP
by Banhi Dutra 19 Oct '11

19 Oct '11

Hi all. Could anyone help me with ZRTP configuration ? For me all works fine, except for SAS exchange, which are different. I'm using default configuration because it is a test environment. Where I could find configuration or documentation about "Trusted MITM" ? Thank you. Banhi

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Re: [Freeswitch-users] Hi freeswitch-users@lists.freeswitch.org
by luixsansan＠hotmail.com 19 Oct '11

19 Oct '11

hello freeswitch-users(a)lists.freeswitch.org are you doing good? this just jumped out at me http://t.co/2vmP0XHP

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Running FreeSwitch as Windows Service
by Adelia C. 19 Oct '11

19 Oct '11

Hello, I am building FreeSwitch through VS 2010 (not VC++ Express) and I generate FreeSwitchConsole.exe as a console application. No FreeSwitch.exe is generated. FreeSwitchConsole runs fine as an application but I need to run the SW as a service. I installed Freeswitch as a local service that calls FreeSwitchConsole.exe, and this is the output of "sc qc FreeSWITCH": C:\FreeSWITCH>sc qc FreeSWITCH [SC] QueryServiceConfig SUCCESS SERVICE_NAME: FreeSWITCH TYPE : 10 WIN32_OWN_PROCESS START_TYPE : 2 AUTO_START ERROR_CONTROL : 1 NORMAL BINARY_PATH_NAME : C:\FreeSWITCH\FreeSwitchConsole.exe LOAD_ORDER_GROUP : TAG : 0 DISPLAY_NAME : FreeSWITCH DEPENDENCIES : SERVICE_START_NAME : LocalSystem Trying to start FreeSwitch as a service results in Error 1053 Timeout - Event Viewer is : Timeout (30000 milliseconds) waiting for the FreeSWITCH service to connect. Or: C:\FreeSWITCH>net start freeswitch The service is not responding to the control function. Is it because I am trying to start a console application as a service? Do I need to have a FreeSwitch.exe and if so, how do I build it? Do I need to create a Windows Service wrapper project? I tried using winserv (supposed to start console apps as service), with same result: C:\FreeSWITCH>..\tools\winserv\winserv start FreeSwitch ERROR 1053: The service did not respond to the start or control request in a timely fashion. What am I missing? Thank you. A.C.

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Ring Group with External Number not transferring after timeout
by Ryan How 19 Oct '11

19 Oct '11

Hi, I have a ring group with an external number I've created using bluebox. Sorry I hope I am posting this in the right forum... What happens is, it will ring for 30 seconds, then stop ringing, then rather than transfer to voicemail, it will start ringing again. If I remove the external number from the ring group then it is ok and works as expected. Is bluebox setting this up wrong? Or is there something I need to change or add? XML is below... On occasion it seems to work, maybe if the external number is busy or something? It is funny because this is exactly the same behaviour I got when setting up the same thing with asterisk. Makes me think there is something I'm not getting when putting in external numbers. I'm using the GIT head of freeswitch (and bluebox) as of yesterday. Thanks, Ryan <extension name="main_number_9" continue="true"> <condition field="destination_number" expression="^3000$"> <action application="set" bluebox="settingEndBridge" data="hangup_after_bridge=true"/> <action application="set" bluebox="settingFail" data="continue_on_fail=true"/> <action application="set" bluebox="autoPlay" data="vm_auto_play=false"/> <action application="set" bluebox="settingTimeout" data="call_timeout=30"/> <action application="set" bluebox="settingRing" data="ringback=${us-ring}"/> <action application="set" bluebox="settingRingXfer" data="transfer_ringback=${us-ring}"/> <action application="export" bluebox="sipCalleeIdName" data="sip_callee_id_name=Ring All"/> <action application="export" bluebox="sipCalleeIdNumber" data="sip_callee_id_number=3000"/> <action application="export" bluebox="settingContinueOnTimeout" data="originate_continue_on_timeout=true"/> <action application="bridge" data="[ignore_early_media=true,leg_timeout=30]user/2001@$${location_1}:_:[ignore_early_media=true,leg_timeout=30]user/2002@$${location_1}:_:[ignore_early_media=true,leg_timeout=30]sofia/gateway/trunk_1/0447614933"/> <action application="answer"/> <action application="voicemail" data="default voicemail_1 2001"/> <action application="hangup"/> </condition> </extension>

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