http://pastebin.freeswitch.org/15500 <http://pastebin.freeswitch.org/15500>Thank you. Malay On Mon, Feb 28, 2011 at 4:55 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
If you read the dialog box closely the login and password are stated in the challenge box. That is our way of telling who pays attention and who doesn't...
not really, its to keep spambots off the site.
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 > >> > > >> > > >> > > >> > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > > >> > >> > >> > >> -- > >> 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 > >> > >> > >> > >> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users@lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > 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 > > > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- 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
UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- 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 UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- 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 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org