[Freeswitch-users] DTMF rfc2833 hanging up the calls

[Mino Haluz]

Hi,

so this is the call log:
http://pastebin.freeswitch.org/21035

the only dialplan I have:
http://pastebin.freeswitch.org/21036

my sip profile:
http://pastebin.freeswitch.org/21038

anything else?

Mino


On Thu, Jun 6, 2013 at 8:12 PM, Steven Ayre <steveayre at 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 at 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 at 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 at 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 at 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 at x.x.x.x:45703 [BREAK]
>>> 2013-06-06 17:13:16.319289 [DEBUG] switch_ivr_bridge.c:533
>>> sofia/myprofile/sip:200 at 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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