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
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
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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
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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
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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
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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
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-- Anthony Minessale II
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AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
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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
http://pastebin.com/CCwmAfZK <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
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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
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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
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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
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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
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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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
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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
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
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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
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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
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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
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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
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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
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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
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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
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
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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
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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
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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
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You are not including the entire log going back to when the call first HITS FS. We need to see the initial invite in the log. On Mon, Feb 28, 2011 at 4:58 PM, Malay Thakershi <mthakershi@gmail.com> wrote:
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
_______________________________________________ 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
I am including everything. First line - until the end. On Mon, Feb 28, 2011 at 5:08 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
You are not including the entire log going back to when the call first HITS FS. We need to see the initial invite in the log.
On Mon, Feb 28, 2011 at 4:58 PM, Malay Thakershi <mthakershi@gmail.com> wrote:
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
_______________________________________________ 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
You did not mention that you are originating the call over event socket or some script. Check your sofia profile xml and make sure you have these set. <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> You are offering every single codec ever at the same time, many of them will not go well with each other in the same offer. On Mon, Feb 28, 2011 at 5:12 PM, Malay Thakershi <mthakershi@gmail.com> wrote:
I am including everything. First line - until the end.
On Mon, Feb 28, 2011 at 5:08 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
You are not including the entire log going back to when the call first HITS FS. We need to see the initial invite in the log.
On Mon, Feb 28, 2011 at 4:58 PM, Malay Thakershi <mthakershi@gmail.com> wrote:
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
_______________________________________________ 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
Ok. I am using managed esl to send originate. I will put these params in sofia profiles. You said I am offering all codecs at once. How do I restrict this? And what should be the codec offered in my case? Thank you. On Feb 28, 2011 5:20 PM, "Anthony Minessale" <anthony.minessale@gmail.com> wrote:
You did not mention that you are originating the call over event socket or some script. Check your sofia profile xml and make sure you have these set.
<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
You are offering every single codec ever at the same time, many of them will not go well with each other in the same offer.
On Mon, Feb 28, 2011 at 5:12 PM, Malay Thakershi <mthakershi@gmail.com> wrote:
I am including everything. First line - until the end.
On Mon, Feb 28, 2011 at 5:08 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
You are not including the entire log going back to when the call first HITS FS. We need to see the initial invite in the log.
On Mon, Feb 28, 2011 at 4:58 PM, Malay Thakershi <mthakershi@gmail.com> wrote:
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 > > 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-- Anthony Minessale II
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AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
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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
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On Mon, Feb 28, 2011 at 4:34 PM, Malay Thakershi <mthakershi@gmail.com>wrote:
Ok. I am using managed esl to send originate.
I will put these params in sofia profiles.
You said I am offering all codecs at once. How do I restrict this? And what should be the codec offered in my case?
Well, assuming that what the other end sent is useful, the other end only offered 3 codecs in the SDP coming back: m=audio 19624 RTP/AVP 0 3 18 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 I'm assuming you don't want the calls to sound horrible, so don't use GSM. If you didn't pay for the G729 commercial licenses then don't use G729. That leaves good ol' PCMU. You can add this to the beginning of your originate dialstring: {absolute_codec_string=PCMU} See if that helps... -MC
OK. So did the following and now I do not get that warning anymore. Also wanted to remove unnecessary codec offerings. 1. Commented out all <!-- Codec Interfaces --> in modules.conf.xml (this reduced comparisons before final codec was being chosen) 2. Changed codec lines in vars.xml <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA"/> 3. Added these to sofia.conf.xml <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> Thank you. Malay On Mon, Feb 28, 2011 at 8:44 PM, Michael Collins <msc@freeswitch.org> wrote:
On Mon, Feb 28, 2011 at 4:34 PM, Malay Thakershi <mthakershi@gmail.com>wrote:
Ok. I am using managed esl to send originate.
I will put these params in sofia profiles.
You said I am offering all codecs at once. How do I restrict this? And what should be the codec offered in my case?
Well, assuming that what the other end sent is useful, the other end only offered 3 codecs in the SDP coming back: m=audio 19624 RTP/AVP 0 3 18 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000
I'm assuming you don't want the calls to sound horrible, so don't use GSM. If you didn't pay for the G729 commercial licenses then don't use G729. That leaves good ol' PCMU. You can add this to the beginning of your originate dialstring: {absolute_codec_string=PCMU}
See if that helps...
-MC
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participants (3)
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Anthony Minessale
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Malay Thakershi
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Michael Collins