[Freeswitch-users] DTMF rfc2833 hanging up the calls

Mon Jun 10 17:36:54 MSD 2013

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 at 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 at 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 at 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 at 192.168.80.55]
>>>
>>>
>>> On Fri, Jun 7, 2013 at 10:57 AM, Mino Haluz <mino.haluz at gmail.com>wrote:
>>>
>>>> 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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>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
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>>>>
>>>
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>>>
>>> 
>>> 
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>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
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>>
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>
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