February 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Monitoring Phone Endpoint Deregistration/Losing Connection
by Avi Marcus 04 Mar '13

04 Mar '13

I'd like to monitor if my user's sip phones are disconnected, to let me know there might be a problem in advance. I see there's a CUSTOM event called sofia::unregister. I presume that triggers whenever a phone doesn't re-register in enough time. However, with default registation of 60 minutes (or even 10) I'd like more granularity. I have the endpoints pinging FS, but I don't see an event or that information stored anywhere. How about setting nat-options-ping (or better, all-reg-options-ping since most NATed endpoint don't actually self-identify). Do I get an event if that fails? I see unregister-on-options-fail but that seemed too aggressive when I tried that in the past. What exactly does the less sever mark-dead-on-options-fail do? Does it remove it from the user/$NUMBER endpoint? Does it trigger an event for me? Other suggestions? -Avi

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Re: [Freeswitch-users] FS with SSL/TLS issues!
by mehroz 01 Mar '13

01 Mar '13

Moreover, i am unable to import "cafile.pem" in my 3CX phone. Could it be the reason if these errors? 1.Do i need client certificate to establish SSL connection with every softphone? 2.I have read that we can use FScomm phone to test but where does it get the certificate file ? 3.How can i check that FS has really seen the agent.pem file and OK for SSL/TLS SIP? Meanwhile i'll map domain name on FS server and will try to generate certificates using domain name. ...and ill be waiting for anyone to put his thoughts in , where i am! Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp7587… Sent from the freeswitch-users mailing list archive at Nabble.com.

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[ERR] switch_ivr_play_say.c:142 Can't find macro
by Archana Venugopan 01 Mar '13

01 Mar '13

Hi , I have created new sounds folder inside en folder and placed new_message_count and I have called this 'new_message_count' in my js script. The wav files inside this macro is being read correctly in development environment whereas in my production which is almost all same as development environment except for the src folder(/usr/srcFreeswitch) does not recognise this 'new_message_count' and throwing this error(2013-02-28 17:37:47.099253 [ERR] switch_ivr_play_say.c:142 Can't find macro new_message_count.) I even gave reload mod_say_en but this 1 is not being picked up. /usr/local/freeswitch/conf/lang/en/vect/sounds.xml <macro name=" new_message_count"> <input pattern="^(.*)$" break_on_match="true"> <match> <action function="play-file" data="voicemail/vectone/vm-welcome1.wav"/> <action function="play-file" data="voicemail/vectone/vm-you_have.wav"/> <action function="say" data="$1" method="pronounced" type="items"/> <action function="play-file" data="voicemail/vectone/vm-newmsgs.wav"/> </match> Can anyone please let me know what went wrong in my other server and why this function is not being picked up. Regards, Archana

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Silence suppression again
by Alex Lake 01 Mar '13

01 Mar '13

Does Freeswitch enable any kind of silence suppression by default? If I have bridge_generate_comfort_noise=true, does that only have an effect when the audio-generating end decides to stop sending RTP? Any other tips for diagnosing silence (it's kind of like a slow noise gate effect)?

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Problem setting up bind_digit_action
by Igor Cappello 01 Mar '13

01 Mar '13

We are experiencing some problems with bind_digit_action. We are trying to get a notification of the digits pressed by the bleg of a bridged call. Our simple test dialplan is the following: <extension name="log_bind_digit_action"> <condition field="destination_number" expression="^MYLOG$"/> <action application="log" data="INFO dtmf ok"/> </extension> <extension name="setup_bind_digit_action"> <condition field="destination_number" expression="^MYSETUP$"/> <action application="bind_digit_action" data="myrealm,*12,exec:execute_extension,MYLOG"/> </extension> <extension name="set binding action"> <condition field="context" expression="public"/> <condition field="destination_number" expression="^test_bind_dtmf$"> <action application="log" data="INFO extension callback"/> <action application="start_dtmf"/> <action application="export" data="execute_on_answer_1=start_dtmf"/>--> <action application="export" data="execute_on_answer_2=execute_extension MYSETUP"/> <action application="digit_action_set_realm" data="myrealm"/> <action application="bridge" data="sofia/gateway/voipcard1/723331"/> <action application="log" data="INFO extension callback . Originate terminated"/> </condition> </extension> We can use only inband DTMF. Here you can find the log of a call (IPs are obscured for security) http://pastebin.com/j8pBFVDL Any hint on why the extension MYLOG is not executed? Maybe we are missing something obvious.. Thanks Igor Cappello

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Freeswitch Installation
by Venkateshwaran Thirugnanam 01 Mar '13

01 Mar '13

Hi All, 1.I need to install freeswitch on centos 6.3 please guild me with procedure..I can't find a procedure on internet...I need to install with Autoboot freeswitch from startup... 2.Why we need to install epel for Centos 6?whether its mandatory or optional? Guild me Regards, Kumaran T

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codec
by Venkateshwaran Thirugnanam 28 Feb '13

28 Feb '13

Hi All, What the difference between Transcodable codecs and Pass-Through Codec? Pls explain Thanks, Kumaran T

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Friday Free For All Topic: GSoC 2013
by Michael Collins 28 Feb '13

28 Feb '13

Hey all! We thought that maybe a good idea for a discussion topic for tomorrow's Friday Free For All (F3A) would be potential projects that FreeSWITCH/OSTAG could sponsor as a mentoring organization. Please bring your fresh ideas and we'll talk about them! Thanks! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Changing codec during calls
by Emrah 28 Feb '13

28 Feb '13

Hi all, How do we go about renegotiating the codecs of a call that is already established? I tried uuid_media_reneg to no avail. E.g.: Upgrade someone who's entered a 16k conference from PCMU to G722. Cheers and thanks, Emrah

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486 Busy Here with Reason: Q.850; cause=16; text="NORMAL_CLEARING"
by Yuriy Nasida 28 Feb '13

28 Feb '13

Hi guys, I see that my FS sends sip message "486 Busy Here" with Reason: Q.850; cause=16;text="NORMAL_CLEARING". Why 16 and "NORMAL_CLEARING ? Probably it should be 17 "user busy". One of my carrier play a "fast busy" (number not available) and blames Q.850 field. Any advice are welcome.Thanks.

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