[Freeswitch-users] disabling ptime warning message

[Anthony Minessale]

look at your SIP traffic and console log.

enter "sofia global siptrace on" followed by "console loglevel debug"
at the cli and make the call.


On Mon, Feb 28, 2011 at 12:47 PM, Malay Thakershi <mthakershi at gmail.com> wrote:
> I have no idea where to look for this setting.
> This is in modules.conf.xml
>     <!-- Codec Interfaces -->
>     <load module="mod_spandsp"/>
>     <!--<load module="mod_voipcodecs"/>-->
>     <load module="mod_g723_1"/>
>     <load module="mod_g729"/>
>     <load module="mod_amr"/>
>     <load module="mod_ilbc"/>
>     <load module="mod_speex"/>
>     <load module="mod_h26x"/>
>     <load module="mod_siren"/>
>     <!--<load module="mod_celt"/>-->
>     <!--<load module="mod_opus"/>-->
> Apart from settings I posted in my previous post, where else to look for
> disabling LPC?
> Malay
> On Mon, Feb 28, 2011 at 12:05 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> is your inbound call using LPC? you don't want to be using LPC and
>> expect anything to sound good that's for sure.
>> It would not just magically say that unless something you are doing has
>> LPC?
>>
>>
>> On Mon, Feb 28, 2011 at 10:24 AM, Malay Thakershi <mthakershi at gmail.com>
>> wrote:
>> > Hello,
>> > I updated to the latest FS version last week.
>> > I started getting the following warning when speech / sound is played on
>> > the
>> > call.
>> > "2011-02-28 10:15:55.985930 [WARNING] sofia_glue.c:213 Codec LPC payload
>> > 7
>> > added to sdp wanting ptime 90 but it's already 20 (G7221:115:20),
>> > disabling
>> > ptime."
>> > I read sections on codecs and negotiations.
>> > Following are the settings from vars.xml (I have not changed them):
>> >   <X-PRE-PROCESS cmd="set"
>> > data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>> >   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>> > Also, there is no codec related setting in sip_profiles files
>> > and sofia.conf.xml file.
>> > I am playing audio files using Cepstral TTS during the call.
>> > Can someone please help me understand these settings? And if they are
>> > appropriate?
>> > Thank you.
>> > Malay
>> > _______________________________________________
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>> > FreeSWITCH-users at lists.freeswitch.org
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>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
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>>
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>>
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>>
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>
>
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>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
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AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
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googletalk:conf+888 at conference.freeswitch.org
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