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Changing DTMF Type for inbound gateway?
by Avi Marcus 01 Mar '13

01 Mar '13

http://wiki.freeswitch.org/wiki/Sofia.conf.xml#DTMF says that you can set a user variable, so e.g. when an inbound gateway ACLs as that user, you can set the dtmf type. As per the wiki, I set: <variable direction="inbound" name="dtmf-type" value="none"/> and do indeed see the variable in my log. However, in the SIP 200 that FS sends back, I still see this in the SDP: rtpmap:101 telephone-event/8000 I tried with sofia loglevel all 9 (I never used that before...) but didn't really see anything related to DTMF. Am I missing something here? -Avi

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dtmf key press issues
by Archana Venugopan 01 Mar '13

01 Mar '13

Hi, I have made few changes to my voicemail flow. Changes were made in mod_voicemail.c,XML files(reload mod_voicemail, reload mod_say_en). After doing the changes if i try to leave a voicemail, the dtmf key press is not being recognised. In front end if I just enable 'Force RFC2833 Out-of-Band DTMF' then my dtmf key press is being recognised. But the issue is if I enable 'Force RFC2833 Out-of-Band DTMF' for outside calls the dtmf key press is not being recognised. Tried changing dtmf-type to inband too but nothing helped. Am not sure whether really my voicemail code changes only affected this though!! Can anyone please help me on this issue. Please let me know why its expecting 'Force RFC2833 Out-of-Band DTMF' to be enabled for dtmf key press recognition and what should be changed so that both my inside and outside key press gets recognised? Many thanks Regards, Archana

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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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