I can't comment on the windows comparability, but it's a one time fee and is only required when you need to transcode (including media bugs for recording, TTS, voicemail, etc.). If you don't do that on a channel then that license isn't used. On Apr 10, 2012 2:02 PM, "Malay Thakershi" <mthakershi@gmail.com> wrote:
Are these licenses supported on windows? If not, how can I use G729 on it?
One license is 1 encoder/decoder per channel. So if I make 30 concurrent calls, do I need 30 licenses?
Is the $10 fee per month or one-time?
If I can't use G729 at all, what could be my options to save bandwidth + get acceptable audio quality?
Thanks for responses.
Malay
On Tue, Apr 10, 2012 at 12:34 AM, Peter Olsson < peter.olsson@visionutveckling.se> wrote:
TTS needs a real license for G729 - please check here how to purchase it: http://www.freeswitch.org/node/235
The free G729 modules is for passthrough only, if you generate audio on the FS instance, it will require encoder/decoder licenses.
/Peter
________________________________ Från: freeswitch-users-bounces@lists.freeswitch.org [ freeswitch-users-bounces@lists.freeswitch.org] för Malay Thakershi [ mthakershi@gmail.com] Skickat: den 10 april 2012 07:22 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Possible to disable core codecs?
Upgraded FS to latest version before coming back for help.
Now, I get error saying: [ERR] mod_g729.c:102 This codec is only usable in passthrough mode! [ERR] switch_core_io.c:1081 Codec G.729 encoder error!
I think what is happening is, it agrees to use G.729 codec but as soon as TTS is opened, there is an error. I read in documentation that mod_g729 is for free codec.
Please help.
Here is the debug trace of the call: ----------------------------------
2012-04-10 00:13:32.940267 [DEBUG] sofia.c:5572 Remote SDP: v=0 o=root 29993 29993 IN IP4 IP1 s=session c=IN IP4 IP1 t=0 0 m=audio 15582 RTP/AVP 0 8 3 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMU:0:8000:20:64000]/[G72 21:115:32000:20:48000]
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMA:8:8000:20:64000]/[G72 21:115:32000:20:48000]
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [GSM:3:8000:20:13200]/[G722 1:115:32000:20:48000]
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[G72 21:115:32000:20:48000]
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [telephone-event:101:8000:2 0:0]/[G7221:115:32000:20:48000]
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMU:0:8000:20:64000]/[AMR :96:8000:20:12200]
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMA:8:8000:20:64000]/[AMR :96:8000:20:12200]
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [GSM:3:8000:20:13200]/[AMR: 96:8000:20:12200]
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[AMR :96:8000:20:12200]
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [telephone-event:101:8000:2 0:0]/[AMR:96:8000:20:12200]
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMU:0:8000:20:64000]/[SPE EX:99:32000:20:44000]
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMA:8:8000:20:64000]/[SPE EX:99:32000:20:44000]
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [GSM:3:8000:20:13200]/[SPEE X:99:32000:20:44000]
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[SPE EX:99:32000:20:44000]
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [telephone-event:101:8000:2 0:0]/[SPEEX:99:32000:20:44000]
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMU:0:8000:20:64000]/[G72 9:18:8000:20:8000]
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMA:8:8000:20:64000]/[G72 9:18:8000:20:8000]
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [GSM:3:8000:20:13200]/[G729 :18:8000:20:8000]
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[G72 9:18:8000:20:8000]
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:3006 Set Codec sofia/sipinterface_1/Phone2@x.24 1.99.201 G729/8000 20 ms 160 samples 8000 bits
2012-04-10 00:13:32.940267 [DEBUG] switch_core_codec.c:111 sofia/sipinterface_1/Phone2@x.241.99 .201 Original read codec set to G729:18
2012-04-10 00:13:32.940267 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/Phone2@ IP1) Running State Change CS_NEW
2012-04-10 00:13:32.940267 [DEBUG] switch_core_state_machine.c:380 (sofia/sipinterface_1/Phone2@ IP1) State NEW
2012-04-10 00:13:32.940267 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101
2012-04-10 00:13:32.940267 [DEBUG] sofia.c:5786 (sofia/sipinterface_1/Phone2@IP1) Stat e Change CS_NEW -> CS_INIT
2012-04-10 00:13:32.940267 [DEBUG] switch_core_session.c:1182 Send signal sofia/sipinterface_1/97278 29132@IP1 [BREAK]
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/Phone2@ IP1) Running State Change CS_INIT
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:401 (sofia/sipinterface_1/Phone2@ IP1) State INIT
2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:85 sofia/sipinterface_1/Phone2@IP1 SOFI A INIT
2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:125 (sofia/sipinterface_1/Phone2@IP1) S tate Change CS_INIT -> CS_ROUTING
2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal sofia/sipinterface_1/97278 29132@IP1 [BREAK]
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:401 (sofia/sipinterface_1/Phone2@ IP1) State INIT going to sleep
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/Phone2@ IP1) Running State Change CS_ROUTING
2012-04-10 00:13:32.960773 [DEBUG] switch_channel.c:1886 (sofia/sipinterface_1/Phone2@x.241.99. 201) Callstate Change DOWN -> RINGING
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410 (sofia/sipinterface_1/Phone2@ IP1) State ROUTING
2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:148 sofia/sipinterface_1/Phone2@IP1 SOF IA ROUTING
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:104 sofia/sipinterface_1/Phone2@6 6.241.99.201 Standard ROUTING
2012-04-10 00:13:32.960773 [INFO] mod_dialplan_xml.c:485 Processing +1Phone2 <Phone2>->Phone1 in context inbound Dialplan: sofia/sipinterface_1/Phone2@IP1 parsing [inbound->vitel-inbound] continue=fa lse Dialplan: sofia/sipinterface_1/Phone2@IP1 Regex (PASS) [vitel-inbound] destination_num ber(8774542559) =~ // break=on-false Dialplan: sofia/sipinterface_1/Phone2@IP1 Action transfer(1056 XML default)
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:154 (sofia/sipinterface_1/Phone2@ IP1) State Change CS_ROUTING -> CS_EXECUTE
2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal sofia/sipinterface_1/97278 29132@IP1 [BREAK]
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410 (sofia/sipinterface_1/Phone2@ IP1) State ROUTING going to sleep
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/Phone2@ IP1) Running State Change CS_EXECUTE
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:417 (sofia/sipinterface_1/Phone2@ IP1) State EXECUTE
2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:241 sofia/sipinterface_1/Phone2@IP1 SOF IA EXECUTE
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:192 sofia/sipinterface_1/Phone2@6 6.241.99.201 Standard EXECUTE EXECUTE sofia/sipinterface_1/Phone2@IP1 transfer(1056 XML default)
2012-04-10 00:13:32.960773 [DEBUG] switch_ivr.c:1711 (sofia/sipinterface_1/Phone2@IP1) State Change CS_EXECUTE -> CS_ROUTING
2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal sofia/sipinterface_1/97278 29132@IP1 [BREAK]
2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:731 Send signal sofia/sipinterface_1/972782 9132@IP1 [BREAK]
2012-04-10 00:13:32.960773 [NOTICE] switch_ivr.c:1717 Transfer sofia/sipinterface_1/Phone2@x.24 1.99.201 to XML[1056@default]
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:417 (sofia/sipinterface_1/Phone2@ IP1) State EXECUTE going to sleep
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/Phone2@ IP1) Running State Change CS_ROUTING
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410 (sofia/sipinterface_1/Phone2@ IP1) State ROUTING
2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:148 sofia/sipinterface_1/Phone2@IP1 SOF IA ROUTING
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:104 sofia/sipinterface_1/Phone2@6 6.241.99.201 Standard ROUTING
2012-04-10 00:13:32.960773 [INFO] mod_dialplan_xml.c:485 Processing +1Phone2 <Phone2>->1056 in context default Dialplan: sofia/sipinterface_1/Phone2@IP1 parsing [default->CHPhoneAsmtDev] continue=f alse Dialplan: sofia/sipinterface_1/Phone2@IP1 Regex (PASS) [CHPhoneAsmtDev] destination_nu mber(1056) =~ /^105\d$/ break=on-false Dialplan: sofia/sipinterface_1/Phone2@IP1 Action sleep(1000) Dialplan: sofia/sipinterface_1/Phone2@IP1 Action managed(clsAsmtApp)
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:154 (sofia/sipinterface_1/Phone2@ IP1) State Change CS_ROUTING -> CS_EXECUTE
2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:1182 Send signal sofia/sipinterface_1/97278 29132@IP1 [BREAK]
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:410 (sofia/sipinterface_1/Phone2@ IP1) State ROUTING going to sleep
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/Phone2@ IP1) Running State Change CS_EXECUTE
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:417 (sofia/sipinterface_1/Phone2@ IP1) State EXECUTE
2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:241 sofia/sipinterface_1/Phone2@IP1 SOF IA EXECUTE
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:192 sofia/sipinterface_1/Phone2@6 6.241.99.201 Standard EXECUTE EXECUTE sofia/sipinterface_1/Phone2@IP1 sleep(1000) EXECUTE sofia/sipinterface_1/Phone2@IP1 managed(clsAsmtApp)
2012-04-10 00:13:33.980239 [DEBUG] switch_cpp.cpp:1227 FreeSWITCH.Managed: attempting to run applica tion 'clsAsmtApp'.
2012-04-10 00:13:34.180422 [DEBUG] switch_cpp.cpp:1172 CoreSession::seHangupHook, hangup_func: 00000 000
2012-04-10 00:13:34.780969 [NOTICE] switch_cpp.cpp:1227 Not an outbound call.
2012-04-10 00:13:34.780969 [INFO] switch_cpp.cpp:1227 caller_id_number: Phone2
2012-04-10 00:13:34.780969 [DEBUG] sofia_glue.c:3258 AUDIO RTP [sofia/sipinterface_1/Phone2@x.2 41.99.201] 10.25.20.202 port 16542 -> IP1 port 15582 codec: 18 ms: 20
2012-04-10 00:13:34.780969 [DEBUG] switch_rtp.c:1669 Not using a timer
2012-04-10 00:13:34.780969 [DEBUG] sofia_glue.c:3522 Set 2833 dtmf send payload to 101
2012-04-10 00:13:34.780969 [DEBUG] sofia_glue.c:3528 Set 2833 dtmf receive payload to 101
2012-04-10 00:13:34.780969 [DEBUG] mod_sofia.c:754 Local SDP sofia/sipinterface_1/Phone2@x.241. 99.201: v=0 o=FreeSWITCH 1334018272 1334018273 IN IP4 64.22.232.56 s=FreeSWITCH c=IN IP4 64.22.232.56 t=0 0 m=audio 16542 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv
2012-04-10 00:13:34.780969 [DEBUG] switch_core_session.c:731 Send signal sofia/sipinterface_1/972782 9132@IP1 [BREAK]
2012-04-10 00:13:34.780969 [DEBUG] switch_channel.c:3244 (sofia/sipinterface_1/Phone2@x.241.99. 201) Callstate Change RINGING -> ACTIVE
2012-04-10 00:13:34.780969 [NOTICE] switch_cpp.cpp:599 Channel [sofia/sipinterface_1/Phone2@x.2 41.99.201] has been answered
2012-04-10 00:13:34.780969 [DEBUG] switch_core_session.c:877 Send signal sofia/sipinterface_1/972782 9132@IP1 [BREAK]
2012-04-10 00:13:34.780969 [DEBUG] sofia.c:5561 Channel sofia/sipinterface_1/Phone2@x.241.99.20 1 entering state [completed][200]
2012-04-10 00:13:34.821006 [DEBUG] switch_core_session.c:877 Send signal sofia/sipinterface_1/972782 9132@IP1 [BREAK]
2012-04-10 00:13:34.821006 [DEBUG] switch_core_session.c:877 Send signal sofia/sipinterface_1/972782 9132@IP1 [BREAK]
2012-04-10 00:13:34.821006 [DEBUG] switch_core_session.c:877 Send signal sofia/sipinterface_1/972782 9132@IP1 [BREAK]
2012-04-10 00:13:35.420577 [INFO] switch_cpp.cpp:1227 OK. Connected to customer DB.
2012-04-10 00:13:35.420577 [DEBUG] switch_ivr_play_say.c:2462 OPEN TTS flite
2012-04-10 00:13:35.420577 [DEBUG] switch_ivr_play_say.c:2471 Raw Codec Activated
2012-04-10 00:13:35.460613 [DEBUG] switch_ivr_play_say.c:2160 Speaking text: <break strength='medium '/>Hello.<break strength='medium'/>Welcome
2012-04-10 00:13:35.481120 [DEBUG] sofia.c:5561 Channel sofia/sipinterface_1/Phone2@x.241.99.20 1 entering state [ready][200]
2012-04-10 00:13:35.481120 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode!
2012-04-10 00:13:35.481120 [ERR] switch_core_io.c:1081 Codec G.729 encoder error!
2012-04-10 00:13:35.500650 [DEBUG] switch_ivr_play_say.c:2354 done speaking text
2012-04-10 00:13:35.500650 [NOTICE] switch_cpp.cpp:1227 ab127cc9-a729-4ea8-b4e0-6863f3ab243f-Inside clsAsmtApp.Run (args ''); HookState is CS_EXECUTE.
2012-04-10 00:13:35.500650 [DEBUG] switch_ivr_play_say.c:2462 OPEN TTS flite
2012-04-10 00:13:35.500650 [DEBUG] switch_ivr_play_say.c:2471 Raw Codec Activated
2012-04-10 00:13:35.520180 [DEBUG] switch_ivr_play_say.c:2160 Speaking text: <break strength='medium '/>We must verify your identity.
2012-04-10 00:13:35.540686 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode!
2012-04-10 00:13:35.540686 [ERR] switch_core_io.c:1081 Codec G.729 encoder error!
2012-04-10 00:13:35.540686 [DEBUG] switch_ivr_play_say.c:2354 done speaking text
2012-04-10 00:13:35.560216 [DEBUG] switch_ivr_play_say.c:1306 Codec Activated L16@8000hz 1 channels 20ms
2012-04-10 00:13:35.560216 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode!
2012-04-10 00:13:35.560216 [ERR] switch_core_io.c:1081 Codec G.729 encoder error! ----------------------
On Thu, Apr 5, 2012 at 1:58 PM, Avi Marcus <avi@avimarcus.net<mailto: avi@avimarcus.net>> wrote: Try absolute_codec_string< http://wiki.freeswitch.org/wiki/Variable_absolute_codec_string> -Avi
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto: 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
!DSPAM:4f83c26e32762080149657!
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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