<html><head><style type='text/css'>p { margin: 0; }</style></head><body><div style='font-family: Arial; font-size: 10pt; color: #000000'>Thanks for the response,<br><br>I went ahead and subscribed to all events, but am still not getting these events. I have also enabled <param name="vad" value="both"/> in my sip profile, not sure if this is redundant or not with the dial plan directives. If VAD is indeed enabled but not working, is there something else I should try? Thanks.<br><hr id="zwchr"><div style="color: rgb(0, 0, 0); font-weight: normal; font-style: normal; text-decoration: none; font-family: Helvetica,Arial,sans-serif; font-size: 12pt;"><b>From: </b>"Michael Collins" <msc@freeswitch.org><br><b>To: </b>"FreeSWITCH Users Help" <freeswitch-users@lists.freeswitch.org><br><b>Sent: </b>Thursday, March 15, 2012 5:29:21 PM<br><b>Subject: </b>Re: [Freeswitch-users] Problem getting TALK and NOTALK events<br><br>Can you confirm if the TALK/NOTALK events are even firing? Try listening to all events and then sifting through the barrage of data to see if those events are present. If they are then you know you just have a filtering issue. If they are not present then you know there's an issue with VAD or something like that.<br>
<br>-MC<br><br><div class="gmail_quote">On Thu, Mar 15, 2012 at 8:59 AM, Dan <span dir="ltr"><<a href="mailto:freeswitch-users@digitaldan.com" target="_blank">freeswitch-users@digitaldan.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Hi, I'm having some issues getting TALK / NOTALK events to fire on an incoming stream to Freeswitch. In my ESL application I am subscribing to RECORD_START RECORD_STOP TALK NOTALK, Below is the dial plan I am using:<br>
<br>
<extension name="teliax-in"><br>
<condition field="destination_number" expression="^(5555551212)$"><br>
<action application="set" data="rtp_enable_vad_in=true"/><br>
<action application="set" data="rtp_enable_vad_out=true"/><br>
<action application="set" data="fire_talk_events=true"/><br>
<action application="set" data="fire_not_talk_events=true"/><br>
<action application="playback" data="ivr/ivr-hello.wav"/><br>
<action application="record" data="/tmp/data.wav"/><br>
</condition><br>
</extension><br>
<br>
I can see that VAD is enabled:<br>
<br>
2012-03-15 09:49:25.523832 [DEBUG] switch_rtp.c:4130 Activate VAD codec PCMU 20ms<br>
2012-03-15 09:49:25.523832 [DEBUG] sofia_glue.c:3353 AUDIO RTP Engage VAD for sofia/external/<a href="mailto:3035551212@10.10.10.1" target="_blank">3035551212@10.10.10.1</a> ( in out )<br>
<br>
<br>
In my ESL app I get the RECORD_START and RECORD_STOP but not the talk events. I'm on git version "2c52f23 2012-02-18 08:37:47 -0600", Any ideas? Thanks.<br>
<br>
<br>
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</blockquote></div><br>
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