[Freeswitch-users] DTMF rfc2833 hanging up the calls

Mon Jun 10 17:11:30 MSD 2013

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