[Freeswitch-users] DTMF rfc2833 hanging up the calls

[Brian Foster]

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