<div dir="ltr">Hi,<div><br></div><div style>so this is the call log:</div><div style><a href="http://pastebin.freeswitch.org/21035">http://pastebin.freeswitch.org/21035</a><br></div><div style><br></div><div style>the only dialplan I have:</div>
<div style><a href="http://pastebin.freeswitch.org/21036">http://pastebin.freeswitch.org/21036</a><br></div><div style><br></div><div style>my sip profile:</div><div style><a href="http://pastebin.freeswitch.org/21038">http://pastebin.freeswitch.org/21038</a><br>
</div><div style><br></div><div style>anything else?</div><div style><br></div><div style>Mino</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, Jun 6, 2013 at 8:12 PM, Steven Ayre <span dir="ltr"><<a href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Can you share a complete debug-level log of the call? Are any other extensions being executed first?<div><br></div><div>
There are api calls that could bind a dtmf key to hanging up the bridge, perhaps that's being done here earlier in the dialplan.</div>
<div><br></div><div>-Steve</div><div class="HOEnZb"><div class="h5"><div><br></div><div><span></span><br><br>On Thursday, June 6, 2013, Mino Haluz wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">FS hangs up the call when it receives RTP DTMF Event.<div><br></div><div>FreeSwitch 1.2.10<div><br></div><div>profile:</div><div><div><profile name="myprofile"></div><div> <settings></div>
<div> <param name="context" value="mycontext"/></div><div> <param name="debug" value="1"/></div><div> <param name="rfc2833-pt" value="101"/></div>
<div> <param name="sip-port" value="5060"/></div><div> <param name="dialplan" value="XML"/></div><div> <param name="dtmf-duration" value="100"/></div>
<div> <param name="dtmf-type" value="rfc2833"/></div><div> <param name="liberal-dtmf" value="true"/></div><div> <param name="codec-ms" value="20"/>--><br>
</div><div> <param name="use-rtp-timer" value="true"/></div><div> <param name="sip-ip" value="192.168.x.x"/></div><div> <param name="rtp-ip" value="192.168.x.x"/></div>
<div> <param name="rtp-timeout-sec" value="3000"/></div><div> <param name="manage-presence" value="false"/><br></div><div> <param name="rtp-timeout-sec" value="5"/><br>
</div><div> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/></div><div> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> </div><div>
</settings><br></div><div></profile></div><div><br></div><div>dialplan:</div><div><div> <extension name="Local_Extension_3"></div><div> <condition field="destination_number" expression="^300$"><br>
</div><div><span style="white-space:pre-wrap"> </span> <action application="bridge" data="sofia/myprofile/300@192.168.x.x:5060"/></div><div> </condition><br></div><div> </extension><br>
</div></div></div><div><br></div><div>modules:</div></div><div><div> <load module="mod_console"/></div><div> <load module="mod_logfile"/></div><div> <load module="mod_enum"/></div>
<div> <load module="mod_event_socket"/></div><div> <load module="mod_sofia"/></div><div> <load module="mod_loopback"/></div><div> <load module="mod_commands"/></div>
<div> <load module="mod_dptools"/></div><div> <load module="mod_expr"/></div><div> <load module="mod_fifo"/></div><div> <load module="mod_hash"/></div>
<div> <load module="mod_amr"/></div><div> <load module="mod_speex"/></div><div> <load module="mod_spandsp"/></div><div> <load module="mod_com_g729"/></div>
<div> <load module="mod_cluechoo"/>--><br></div><div> <load module="mod_dialplan_xml"/></div><div> <load module="mod_g723_1"/></div><div> <load module="mod_native_file"/></div>
<div> <load module="mod_local_stream"/></div><div> <load module="mod_tone_stream"/></div><div> <load module="mod_xml_cdr"/></div><div><br></div></div></div><div class="gmail_extra">
<br><br><div class="gmail_quote">On Thu, Jun 6, 2013 at 5:20 PM, Mino Haluz <span dir="ltr"><<a>mino.haluz@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">Hi,<div><br></div><div>if I use SIP INFO on leg A, FS detects everything fine, but in case I use rfc2833, after pressing number, it instantly hangs up the call.</div><div><br></div><div><div>
n2013-06-06 17:13:16.319289 [NOTICE] mod_sofia.c:1137 Hangup sofia/myprofile/sip:200@x.x.x.x:45703 [CS_EXCHANGE_MEDIA] [MEDIA_TIMEOUT]</div><div>2013-06-06 17:13:16.319289 [DEBUG] switch_channel.c:3096 Send signal sofia/myprofile/sip:200@x.x.x.x:45703 [KILL]</div>
<div>2013-06-06 17:13:16.319289 [DEBUG] switch_core_session.c:1333 Send signal sofia/myprofile/sip:200@x.x.x.x:45703 [BREAK]</div><div>2013-06-06 17:13:16.319289 [DEBUG] switch_ivr_bridge.c:533 sofia/myprofile/sip:200@x.x.x.x:45703 ending bridge by request from read function</div>
<div><br></div><div>I do not use any ivr module, or start_dtmf, or whatsoever setting related to dtmf. Any hint?</div><div><br></div><div>Thanks, </div><span><font color="#888888"><div>Mino</div></font></span></div>
</div>
</blockquote></div><br></div>
</blockquote></div>
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