On Mon, Feb 28, 2011 at 4:53 PM, Malay Thakershi <
mthakershi@gmail.com> wrote:
> I can't login to FS pastebin.
> I used same user / password as I use to access users group.
> Malay
>
> On Mon, Feb 28, 2011 at 3:47 PM, Anthony Minessale
> <
anthony.minessale@gmail.com> wrote:
>>
>> You also need "sofia global siptrace on" so we can see the SIP traffic
>> in the log.
>> We have our own pastebin at
http://pastebin.freeswitch.org
>>
>>
>> On Mon, Feb 28, 2011 at 3:22 PM, Malay Thakershi <
mthakershi@gmail.com>
>> wrote:
>> >
http://pastebin.com/CCwmAfZK
>> > I noticed that LPC warning only comes for outbound calls I initiate
>> > using
>> > "Originate" API.
>> > Malay
>> >
>> > On Mon, Feb 28, 2011 at 2:04 PM, Anthony Minessale
>> > <
anthony.minessale@gmail.com> wrote:
>> >>
>> >> That's L16 not LPC, you need L16 to play files.
>> >> Why don't you just put the whole log of the call on pastebin.
>> >>
>> >>
>> >> On Mon, Feb 28, 2011 at 1:53 PM, Malay Thakershi <
mthakershi@gmail.com>
>> >> wrote:
>> >> > Following is a section from the log:
>> >> > ---------------------------
>> >> > 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
>> >> >> > <!-- 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@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
>> >> >> >> >
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >> >
>> >> >> >> >
>> >> >> >> >
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>> >> >> >> >
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>> >> >> >> >
>> >> >> >> >
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> --
>> >> >> >> Anthony Minessale II
>> >> >> >>
>> >> >> >> FreeSWITCH
http://www.freeswitch.org/
>> >> >> >> ClueCon
http://www.cluecon.com/
>> >> >> >> Twitter:
http://twitter.com/FreeSWITCH_wire
>> >> >> >>
>> >> >> >> AIM: anthm
>> >> >> >>
MSN:anthony_minessale@hotmail.com
>> >> >> >> GTALK/JABBER/
PAYPAL:anthony.minessale@gmail.com
>> >> >> >> IRC:
irc.freenode.net #freeswitch
>> >> >> >>
>> >> >> >> FreeSWITCH Developer Conference
>> >> >> >>
sip:888@conference.freeswitch.org
>> >> >> >>
googletalk:conf+888@conference.freeswitch.org
>> >> >> >> pstn:+19193869900
>> >> >> >>
>> >> >> >> _______________________________________________
>> >> >> >> FreeSWITCH-users mailing list
>> >> >> >>
FreeSWITCH-users@lists.freeswitch.org
>> >> >> >>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >>
>> >> >> >>
>> >> >> >>
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>> >> >> >>
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>> >> >> >
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> >
FreeSWITCH-users@lists.freeswitch.org
>> >> >> >
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> >
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>> >> >> >
>> >> >> >
>> >> >>
>> >> >>
>> >> >>
>> >> >> --
>> >> >> Anthony Minessale II
>> >> >>
>> >> >> FreeSWITCH
http://www.freeswitch.org/
>> >> >> ClueCon
http://www.cluecon.com/
>> >> >> Twitter:
http://twitter.com/FreeSWITCH_wire
>> >> >>
>> >> >> AIM: anthm
>> >> >>
MSN:anthony_minessale@hotmail.com
>> >> >> GTALK/JABBER/
PAYPAL:anthony.minessale@gmail.com
>> >> >> IRC:
irc.freenode.net #freeswitch
>> >> >>
>> >> >> FreeSWITCH Developer Conference
>> >> >>
sip:888@conference.freeswitch.org
>> >> >>
googletalk:conf+888@conference.freeswitch.org
>> >> >> pstn:+19193869900
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >>
FreeSWITCH-users@lists.freeswitch.org
>> >> >>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
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>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> >
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>> >> >
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>> >> >
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>> >> >
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>> >> >
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH
http://www.freeswitch.org/
>> >> ClueCon
http://www.cluecon.com/
>> >> Twitter:
http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >>
MSN:anthony_minessale@hotmail.com
>> >> GTALK/JABBER/
PAYPAL:anthony.minessale@gmail.com
>> >> IRC:
irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >>
sip:888@conference.freeswitch.org
>> >>
googletalk:conf+888@conference.freeswitch.org
>> >> pstn:+19193869900
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >>
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>> >>
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>> >>
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>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> >
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>> >
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>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH
http://www.freeswitch.org/
>> ClueCon
http://www.cluecon.com/
>> Twitter:
http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>>
MSN:anthony_minessale@hotmail.com
>> GTALK/JABBER/
PAYPAL:anthony.minessale@gmail.com
>> IRC:
irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>>
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>>
googletalk:conf+888@conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
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>
>
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>
--
Anthony Minessale II
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AIM: anthm
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IRC:
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FreeSWITCH Developer Conference
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