You shouldn't need to set that. Are you sure it's not caused by the low RTP timeout of 5 seconds on the profile?

/Peter


2013/6/10 Mino Haluz <mino.haluz@gmail.com>
Ok timer_name = soft did it and now it works. Can someone elaborate what this value means in fact?


On Mon, Jun 10, 2013 at 3:11 PM, Mino Haluz <mino.haluz@gmail.com> wrote:
Complete log with siptrace on: http://pastebin.freeswitch.org/21046


On Mon, Jun 10, 2013 at 6:51 AM, Brian Foster <bdfoster@davri.com> wrote:

Need a siptrace, please run 'sofia global siptrace on' before you take the console log and post it on pastebin.freeswitch.org. Reply back with the link. Hopefully we can glean some more info.

- BDF

On Jun 7, 2013 5:09 AM, "Mino Haluz" <mino.haluz@gmail.com> wrote:
I forgot to delete all sofia_reg lines (registrations made a bit mess in call log). So for example this line should not be there:

2013-06-07 10:47:31.919290 [DEBUG] sofia_reg.c:1520 Send challenge for [300@192.168.80.55]


On Fri, Jun 7, 2013 at 10:57 AM, Mino Haluz <mino.haluz@gmail.com> wrote:
Hi,

so this is the call log:

the only dialplan I have:

my sip profile:

anything else?

Mino


On Thu, Jun 6, 2013 at 8:12 PM, Steven Ayre <steveayre@gmail.com> wrote:
Can you share a complete debug-level log of the call? Are any other extensions being executed first?

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.

-Steve



On Thursday, June 6, 2013, Mino Haluz wrote:
FS hangs up the call when it receives RTP DTMF Event.

FreeSwitch 1.2.10

profile:
<profile name="myprofile">
    <settings>
  <param name="context" value="mycontext"/>
  <param name="debug" value="1"/>
  <param name="rfc2833-pt" value="101"/>
  <param name="sip-port" value="5060"/>
  <param name="dialplan" value="XML"/>
  <param name="dtmf-duration" value="100"/>
  <param name="dtmf-type" value="rfc2833"/>
  <param name="liberal-dtmf" value="true"/>
  <param name="codec-ms" value="20"/>-->
  <param name="use-rtp-timer" value="true"/>
  <param name="sip-ip" value="192.168.x.x"/>
  <param name="rtp-ip" value="192.168.x.x"/>
  <param name="rtp-timeout-sec" value="3000"/>
  <param name="manage-presence" value="false"/>
  <param name="rtp-timeout-sec" value="5"/>
  <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
  <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> 
    </settings>
</profile>

dialplan:
    <extension name="Local_Extension_3">
      <condition field="destination_number" expression="^300$">
   <action application="bridge" data="sofia/myprofile/300@192.168.x.x:5060"/>
      </condition>
    </extension>

modules:
    <load module="mod_console"/>
    <load module="mod_logfile"/>
    <load module="mod_enum"/>
    <load module="mod_event_socket"/>
    <load module="mod_sofia"/>
    <load module="mod_loopback"/>
    <load module="mod_commands"/>
    <load module="mod_dptools"/>
    <load module="mod_expr"/>
    <load module="mod_fifo"/>
    <load module="mod_hash"/>
    <load module="mod_amr"/>
    <load module="mod_speex"/>
    <load module="mod_spandsp"/>
    <load module="mod_com_g729"/>
    <load module="mod_cluechoo"/>-->
    <load module="mod_dialplan_xml"/>
    <load module="mod_g723_1"/>
    <load module="mod_native_file"/>
    <load module="mod_local_stream"/>
    <load module="mod_tone_stream"/>
    <load module="mod_xml_cdr"/>



On Thu, Jun 6, 2013 at 5:20 PM, Mino Haluz <mino.haluz@gmail.com> wrote:
Hi,

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.

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]
2013-06-06 17:13:16.319289 [DEBUG] switch_channel.c:3096 Send signal sofia/myprofile/sip:200@x.x.x.x:45703 [KILL]
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]
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

I do not use any ivr module, or start_dtmf, or whatsoever setting related to dtmf. Any hint?

Thanks, 
Mino


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_________________________________________________________________________
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
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http://wiki.freeswitch.org
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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