[Freeswitch-users] no RTP send during Voice Mail recording

kokoska rokoska kokoska.rokoska at post.cz
Sat Apr 25 13:56:30 PDT 2009


Thank you very much, Dave, for your help!

Mentioned modification did the trick and all works as I wish :-) Thank you!

BTW: When I look into the pcap files for RTP stream, I see some strange
timing for outgoing RTP from FreeSWITCH. Every even packet is sent after
24 ms and every odd packet at 12 ms...
"ptime" is on both sides set to 20 ms, incomming stream has nearly no
jitter, server is on real HW (no virtualization) and almost idle (the
most consuming process is "htop" :-)
Is it normal or should I investigate what is wrong?

Best regards,

kokoska.rokoska


David Knell napsal(a):
> Add something like
> memset(write_buf, 0, SWITCH_RECOMMENDED_BUFFER_SIZE);
> after 
> char write_buf[SWITCH_RECOMMENDED_BUFFER_SIZE];
> in switch_ivr_play_say.c (line 395)
> 
> --Dave
> 
>> Thank you very much, dujinfang, for your help!
>>
>> When I use
>> <action application="set" data="record_waste_resources=true"/>
>>
>> the FreeSWITCH really sends back RTP stream during recording, but
>> instead of (faked) silence it is full of completely regular load noise :-)
>> I have tested it with different devices (Linskys, Snom, FritzBOX,
>> Nokia...) with the same result (even pcap files looks similar).
>>
>> Dialplan snipped looks like:
>> <action application='answer'/>
>> <action application='playback' data='silence_stream://1000'/>
>> <action application='set' data='record_waste_resources=true'/>
>> <action application='voicemail' data='context $${domain} number'/>
>> <action application='hangup'/>
>>
>> Do you (or anybody else :-) know what I'm doing wrong?
>>
>> Thanks once more, dujinfang, for your help!
>>
>> Best regards,
>>
>> kokoska.rokoska
>>
>>
>>
>> dujinfang napsal(a):
>>> I haven't tested but I guess it's just like other variables and I  
>>> documented to here:
>>>
>>> http://wiki.freeswitch.org/wiki/Channel_Variables#record_waste_resources
>>>
>>>
>>> On Apr 25, 2009, at 11:56 PM, kokoska rokoska wrote:
>>>
>>>> Thank you very much, Anthony, for such fast solution!
>>>>
>>>> May I ask you - How should I activate this feature?
>>>> I have tried to "grep" through sources for new NDLB variable but I
>>>> didn't find one...
>>>>
>>>> Best regards,
>>>>
>>>> kokoska.rokoska
>>>>
>>>> Anthony Minessale napsal(a):
>>>>> sigh,
>>>>>
>>>>> see r13144
>>>>>
>>>>>
>>>>> On Fri, Apr 24, 2009 at 3:43 AM, kokoska rokoska
>>>>> <kokoska.rokoska at post.cz <mailto:kokoska.rokoska at post.cz>> wrote:
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>    seven napsal(a):
>>>>>> You are not alone, I vote 1.
>>>>>>
>>>>>> And there's a similer variable in conference:
>>>>>>
>>>>>>       <!--Can be | delim of waste|mute|deaf waste will always
>>>>>> transmit data to each channel
>>>>>>           even during silence -->
>>>>>>       <!--<param name="member-flags" value="waste"/>-->
>>>>>>
>>>>>    Thank you very much, seven, for your support :-)
>>>>>
>>>>>    Best regards,
>>>>>
>>>>>    kokoska.rokoska
>>>>>
>>>>>
>>>>>    _______________________________________________
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>>>>>    http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> -- 
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
>>>>> ClueCon http://www.cluecon.com/
>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com
>>>>> <mailto:MSN%3Aanthony_minessale at hotmail.com>
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>>>>> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org
>>>>> <mailto:sip%3A888 at conference.freeswitch.org>
>>>>> iax:guest at conference.freeswitch.org/888
>>>>> <http://iax:guest@conference.freeswitch.org/888>
>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>>> pstn:213-799-1400
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
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> 
> 
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