[Freeswitch-users] DTMF rfc2833 hanging up the calls

Mon Jun 10 18:19:39 MSD 2013

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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>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
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>>>>>>
>>>>>
>>>>
>>>>
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>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
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>>
>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
> 
> 
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