[Freeswitch-users] DTMF rfc2833 hanging up the calls

Mino Haluz mino.haluz at gmail.com
Mon Jun 10 18:47:04 MSD 2013 

DTMF tone lasted 100ms, so I am sure this timer could not be hit.. When I
delete the soft timer, it does not work. Maybe it's something with -
http://wiki.freeswitch.org/wiki/RTP_Issues


On Mon, Jun 10, 2013 at 4:19 PM, Peter Olsson <peter at olssononline.se> wrote:

> 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 at 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 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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>>>>>>
>>>>>
>>>>>
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>>>>
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>>>>
>>>> 
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>>
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>>
>> 
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>
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
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