2011-02-28 13:36:08.294241 [DEBUG] sofia_glue.c:2760 Set Codec sofia/sipinterface_1/##########@##.##.##.## PCMU/8000 20 ms 160 samples 64000 bits

2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2298 OPEN TTS cepstral

2011-02-28 13:36:15.015491 [DEBUG] switch_ivr_play_say.c:2307 Raw Codec Activated

2011-02-28 13:36:15.017444 [DEBUG] switch_ivr_play_say.c:1996 Speaking text: <break strength='medium'/>We must verify your identity.

2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:2188 done speaking text

2011-02-28 13:36:20.295426 [DEBUG] switch_ivr_play_say.c:1244 Codec Activated L16@8000hz 1 channels 20ms

2011-02-28 13:36:25.530443 [DEBUG] switch_ivr_play_say.c:1581 done playing file

2011-02-28 13:36:25.530443 [DEBUG] switch_ivr.c:1135 Codec Activated L16@8000hz 1 channels 20ms

2011-02-28 13:36:30.549653 [DEBUG] switch_ivr_play_say.c:1244 Codec Activated L16@8000hz 1 channels 20ms

---------------------------

So I see when I play a file (using StreamFile / PAGD), it activates L16, which the wiki pages says is not recommended. So should I deactivate it? If so, how?

Now, I have not done any setting out of default / ordinary that comes with the build. I am playing WAV file that is generated by Cepstral SWIFT command line tool (text to WAV). The file format is "Wave PCM signed 16 bit, 8000 Hz, 128 kbps, mono".

Thank you for help so far.

Malay

On Mon, Feb 28, 2011 at 12:50 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:

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@gmail.com> wrote:
> I have no idea where to look for this setting.
> This is in modules.conf.xml
>    
>    <load module="mod_spandsp"/>
>    
>    <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"/>
>    
>    
> 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@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@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@32000h,G7221@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
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users@lists.freeswitch.org
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>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
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>>
>> AIM: anthm
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>>
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>>
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--
Anthony Minessale II

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