Try absolute_codec_string<http://wiki.freeswitch.org/wiki/Variable_absolute_codec_string> -Avi

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!

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

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True, but sufficient demand could change that. :-) Steve On 04/11/2012 03:15 AM, Michael Collins wrote:

Sorry, no g729 licenses on non-Linux plaforms. (Sorry Mac, BSD, Windows users.) -MC

On Tue, Apr 10, 2012 at 11:16 AM, Brian Foster <bdfoster@endigotech.com <mailto:bdfoster@endigotech.com>> wrote:

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.

Thanks for information. But I am not able to use any other codec than PCMU/PCMA for some reason. I read related settings and what they mean. I am using Vitelity for now and they say they do support G722/GSM/G729 on their end. In sip_profiles, I have <param name="inbound-codec-negotiation" value="greedy"/> (tried scrooge but no success) In vars.xml, I have <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,GSM,PCMU,PCMA"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,GSM,PCMU,PCMA"/> In modules.conf.xml, I have <load module="mod_spandsp"/> Debug trace from console: ----- 2012-04-10 18:00:17.430961 [DEBUG] sofia.c:7547 IP IP1 Approved by acl "Malay_ACL[]". Access Granted. 2012-04-10 18:00:17.430961 [NOTICE] switch_channel.c:926 New Channel sofia/sipinterface_1/7607058888@IP2 [45fe1639-ead1-4547-b063-e40f38b5fd04] 2012-04-10 18:00:17.430961 [DEBUG] sofia.c:5561 Channel sofia/sipinterface_1/7607058888@IP2 entering state [received][100] 2012-04-10 18:00:17.430961 [DEBUG] sofia.c:5572 Remote SDP: v=0 o=root 15681 15681 IN IP4 IP2 s=session c=IN IP4 IP2 t=0 0 m=audio 12860 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 18:00:17.430961 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/7607058888@IP2) Running State Change CS_NEW 2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMU:0:8000:20:64000]/[LPC:7:8000:90:2400] 2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMA:8:8000:20:64000]/[LPC:7:8000:90:2400] 2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [GSM:3:8000:20:13200]/[LPC:7:8000:90:2400] 2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:380 (sofia/sipinterface_1/7607058888@IP2) State NEW 2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[LPC:7:8000:90:2400] 2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [telephone-event:101:8000:20:0]/[LPC:7:8000:90:2400] 2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101 2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMU:0:8000:20:64000]/[G726-16:124:8000:20:16000] 2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMA:8:8000:20:64000]/[G726-16:124:8000:20:16000] 2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [GSM:3:8000:20:13200]/[G726-16:124:8000:20:16000] 2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[G726-16:124:8000:20:16000] 2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [telephone-event:101:8000:20:0]/[G726-16:124:8000:20:16000] 2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101 2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMU:0:8000:20:64000]/[AAL2-G726-16:124:8000:20:16000] 2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMA:8:8000:20:64000]/[AAL2-G726-16:124:8000:20:16000] 2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [GSM:3:8000:20:13200]/[AAL2-G726-16:124:8000:20:16000] 2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [G729:18:8000:20:8000]/[AAL2-G726-16:124:8000:20:16000] 2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [telephone-event:101:8000:20:0]/[AAL2-G726-16:124:8000:20:16000] 2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101 2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMA:8:8000:20:64000] 2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4895 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] 2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:4911 Bah HUMBUG! Sticking with PCMA@8000h@20i 2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:3006 Set Codec sofia/sipinterface_1/7607058888@IP2 PCMA/8000 20 ms 160 samples 64000 bits 2012-04-10 18:00:17.430961 [DEBUG] switch_core_codec.c:111 sofia/sipinterface_1/7607058888@IP2 Original read codec set to PCMA:8 2012-04-10 18:00:17.430961 [DEBUG] sofia_glue.c:5016 Set 2833 dtmf send/recv payload to 101 2012-04-10 18:00:17.430961 [DEBUG] sofia.c:5786 (sofia/sipinterface_1/7607058888@IP2) State Change CS_NEW -> CS_INIT 2012-04-10 18:00:17.430961 [DEBUG] switch_core_session.c:1182 Send signal sofia/sipinterface_1/7607058888@IP2 [BREAK] 2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/7607058888@IP2) Running State Change CS_INIT 2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:401 (sofia/sipinterface_1/7607058888@IP2) State INIT 2012-04-10 18:00:17.430961 [DEBUG] mod_sofia.c:85 sofia/sipinterface_1/7607058888@IP2 SOFIA INIT 2012-04-10 18:00:17.430961 [DEBUG] mod_sofia.c:125 (sofia/sipinterface_1/7607058888@IP2) State Change CS_INIT -> CS_ROUTING 2012-04-10 18:00:17.430961 [DEBUG] switch_core_session.c:1182 Send signal sofia/sipinterface_1/7607058888@IP2 [BREAK] 2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:401 (sofia/sipinterface_1/7607058888@IP2) State INIT going to sleep 2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/7607058888@IP2) Running State Change CS_ROUTING 2012-04-10 18:00:17.430961 [DEBUG] switch_channel.c:1886 (sofia/sipinterface_1/7607058888@IP2) Callstate Change DOWN -> RINGING 2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:410 (sofia/sipinterface_1/7607058888@IP2) State ROUTING 2012-04-10 18:00:17.430961 [DEBUG] mod_sofia.c:148 sofia/sipinterface_1/7607058888@IP2 SOFIA ROUTING 2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:104 sofia/sipinterface_1/7607058888@IP2 Standard ROUTING 2012-04-10 18:00:17.430961 [INFO] mod_dialplan_xml.c:485 Processing +17607058888 <7607058888>->DID1 in context inbound Dialplan: sofia/sipinterface_1/7607058888@IP2 parsing [inbound->vitel-inbound] continue=false Dialplan: sofia/sipinterface_1/7607058888@IP2 Regex (PASS) [vitel-inbound] destination_number(DID1) =~ // break=on-false Dialplan: sofia/sipinterface_1/7607058888@IP2 Action transfer(1056 XML default) 2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:154 (sofia/sipinterface_1/7607058888@IP2) State Change CS_ROUTING -> CS_EXECUTE 2012-04-10 18:00:17.430961 [DEBUG] switch_core_session.c:1182 Send signal sofia/sipinterface_1/7607058888@IP2 [BREAK] 2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:410 (sofia/sipinterface_1/7607058888@IP2) State ROUTING going to sleep 2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/7607058888@IP2) Running State Change CS_EXECUTE 2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:417 (sofia/sipinterface_1/7607058888@IP2) State EXECUTE 2012-04-10 18:00:17.430961 [DEBUG] mod_sofia.c:241 sofia/sipinterface_1/7607058888@IP2 SOFIA EXECUTE 2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:192 sofia/sipinterface_1/7607058888@IP2 Standard EXECUTE EXECUTE sofia/sipinterface_1/7607058888@IP2 transfer(1056 XML default) 2012-04-10 18:00:17.430961 [DEBUG] switch_ivr.c:1711 (sofia/sipinterface_1/7607058888@IP2) State Change CS_EXECUTE -> CS_ROUTING 2012-04-10 18:00:17.430961 [DEBUG] switch_core_session.c:1182 Send signal sofia/sipinterface_1/7607058888@IP2 [BREAK] 2012-04-10 18:00:17.430961 [DEBUG] switch_core_session.c:731 Send signal sofia/sipinterface_1/7607058888@IP2 [BREAK] 2012-04-10 18:00:17.430961 [NOTICE] switch_ivr.c:1717 Transfer sofia/sipinterface_1/7607058888@IP2 to XML[1056@default] 2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:417 (sofia/sipinterface_1/7607058888@IP2) State EXECUTE going to sleep 2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/7607058888@IP2) Running State Change CS_ROUTING 2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:410 (sofia/sipinterface_1/7607058888@IP2) State ROUTING 2012-04-10 18:00:17.430961 [DEBUG] mod_sofia.c:148 sofia/sipinterface_1/7607058888@IP2 SOFIA ROUTING 2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:104 sofia/sipinterface_1/7607058888@IP2 Standard ROUTING 2012-04-10 18:00:17.430961 [INFO] mod_dialplan_xml.c:485 Processing +17607058888 <7607058888>->1056 in context default Dialplan: sofia/sipinterface_1/7607058888@IP2 parsing [default->CHPhoneAsmtDev] continue=false Dialplan: sofia/sipinterface_1/7607058888@IP2 Regex (PASS) [CHPhoneAsmtDev] destination_number(1056) =~ /^105\d$/ break=on-false Dialplan: sofia/sipinterface_1/7607058888@IP2 Action sleep(1000) Dialplan: sofia/sipinterface_1/7607058888@IP2 Action managed(clsAsmtApp) 2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:154 (sofia/sipinterface_1/7607058888@IP2) State Change CS_ROUTING -> CS_EXECUTE 2012-04-10 18:00:17.430961 [DEBUG] switch_core_session.c:1182 Send signal sofia/sipinterface_1/7607058888@IP2 [BREAK] 2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:410 (sofia/sipinterface_1/7607058888@IP2) State ROUTING going to sleep 2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/7607058888@IP2) Running State Change CS_EXECUTE 2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:417 (sofia/sipinterface_1/7607058888@IP2) State EXECUTE 2012-04-10 18:00:17.430961 [DEBUG] mod_sofia.c:241 sofia/sipinterface_1/7607058888@IP2 SOFIA EXECUTE 2012-04-10 18:00:17.430961 [DEBUG] switch_core_state_machine.c:192 sofia/sipinterface_1/7607058888@IP2 Standard EXECUTE EXECUTE sofia/sipinterface_1/7607058888@IP2 sleep(1000) EXECUTE sofia/sipinterface_1/7607058888@IP2 managed(clsAsmtApp) 2012-04-10 18:00:18.451403 [DEBUG] switch_cpp.cpp:1227 FreeSWITCH.Managed: attempting to run application 'clsAsmtApp'. 2012-04-10 18:00:18.650609 [DEBUG] switch_cpp.cpp:1172 CoreSession::seHangupHook, hangup_func: 00000000 2012-04-10 18:00:19.491376 [NOTICE] switch_cpp.cpp:1227 Not an outbound call. 2012-04-10 18:00:19.491376 [INFO] switch_cpp.cpp:1227 caller_id_number: 7607058888 2012-04-10 18:00:19.491376 [DEBUG] sofia_glue.c:3258 AUDIO RTP [sofia/sipinterface_1/7607058888@IP2] 10.25.20.202 port 31458 -> IP2 port 12 odec: 8 ms: 20 2012-04-10 18:00:19.491376 [DEBUG] switch_rtp.c:1669 Not using a timer 2012-04-10 18:00:19.510906 [DEBUG] sofia_glue.c:3522 Set 2833 dtmf send payload to 101 2012-04-10 18:00:19.510906 [DEBUG] sofia_glue.c:3528 Set 2833 dtmf receive payload to 101 2012-04-10 18:00:19.510906 [DEBUG] mod_sofia.c:754 Local SDP sofia/sipinterface_1/7607058888@IP2: v=0 o=FreeSWITCH 1334067361 1334067362 IN IP4 64.22.232.56 s=FreeSWITCH c=IN IP4 64.22.232.56 t=0 0 m=audio 31458 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2012-04-10 18:00:19.510906 [DEBUG] switch_core_session.c:731 Send signal sofia/sipinterface_1/7607058888@IP2 [BREAK] 2012-04-10 18:00:19.510906 [DEBUG] switch_channel.c:3244 (sofia/sipinterface_1/7607058888@IP2) Callstate Change RINGING -> ACTIVE 2012-04-10 18:00:19.510906 [DEBUG] switch_core_session.c:877 Send signal sofia/sipinterface_1/7607058888@IP2 [BREAK] 2012-04-10 18:00:19.510906 [NOTICE] switch_cpp.cpp:599 Channel [sofia/sipinterface_1/7607058888@IP2] has been answered 2012-04-10 18:00:19.510906 [DEBUG] sofia.c:5561 Channel sofia/sipinterface_1/7607058888@IP2 entering state [completed][200] 2012-04-10 18:00:19.550942 [DEBUG] switch_core_session.c:877 Send signal sofia/sipinterface_1/7607058888@IP2 [BREAK] 2012-04-10 18:00:19.550942 [DEBUG] switch_core_session.c:877 Send signal sofia/sipinterface_1/7607058888@IP2 [BREAK] 2012-04-10 18:00:19.550942 [DEBUG] switch_core_session.c:877 Send signal sofia/sipinterface_1/7607058888@IP2 [BREAK] 2012-04-10 18:00:20.331166 [INFO] switch_cpp.cpp:1227 OK. Connected to customer DB. 2012-04-10 18:00:20.331166 [DEBUG] switch_ivr_play_say.c:2462 OPEN TTS flite 2012-04-10 18:00:20.331166 [DEBUG] switch_ivr_play_say.c:2471 Raw Codec Activated 2012-04-10 18:00:20.390732 [DEBUG] switch_ivr_play_say.c:2160 Speaking text: <break strength='medium'/>Hello.<break strength='medium'/>Welcome. 2012-04-10 18:00:20.411239 [DEBUG] sofia.c:5561 Channel sofia/sipinterface_1/7607058888@IP2 entering state [ready][200] 2012-04-10 18:00:20.611421 [DEBUG] switch_rtp.c:3224 Correct ip/port confirmed. 2012-04-10 18:00:26.071033 [DEBUG] switch_core_session.c:877 Send signal sofia/sipinterface_1/7607058888@IP2 [BREAK] 2012-04-10 18:00:26.071033 [DEBUG] switch_channel.c:2848 (sofia/sipinterface_1/7607058888@IP2) Callstate Change ACTIVE -> HANGUP 2012-04-10 18:00:26.071033 [NOTICE] sofia.c:636 Hangup sofia/sipinterface_1/7607058888@IP2 [CS_EXECUTE] [NORMAL_CLEARING] 2012-04-10 18:00:26.071033 [DEBUG] switch_channel.c:2871 Send signal sofia/sipinterface_1/7607058888@IP2 [KILL] 2012-04-10 18:00:26.071033 [DEBUG] switch_cpp.cpp:1227 AppFunction is in hangupCallback. 2012-04-10 18:00:26.071033 [NOTICE] switch_cpp.cpp:1227 45fe1639-ead1-4547-b063-e40f38b5fd04-clsAsmtApp hanging up, UUID: 45fe1639-ead1-4547-b063-e40f38b5f 2012-04-10 18:00:26.071033 [DEBUG] switch_core_session.c:1182 Send signal sofia/sipinterface_1/7607058888@IP2 [BREAK] 2012-04-10 18:00:26.071033 [DEBUG] switch_ivr_play_say.c:2354 done speaking text 2012-04-10 18:00:26.071033 [NOTICE] switch_cpp.cpp:1227 45fe1639-ead1-4547-b063-e40f38b5fd04-Inside clsAsmtApp.Run (args ''); HookState is CS_HANGUP. 2012-04-10 18:00:26.091539 [DEBUG] switch_cpp.cpp:635 CoreSession::hangup 2012-04-10 18:00:26.231179 [INFO] switch_cpp.cpp:1227 Entered 6 SpeakDelayReport in ThCustomerDb database. 2012-04-10 18:00:26.231179 [INFO] switch_cpp.cpp:1227 Entered 2 PathDelayReport in ThCustomerDb database. 2012-04-10 18:00:26.250709 [INFO] switch_cpp.cpp:1227 Entered 4 IVRAsmtStatusRecord in ThCustomerDb database. 2012-04-10 18:00:26.250709 [DEBUG] switch_cpp.cpp:1007 sofia/sipinterface_1/7607058888@IP2 destroy/unlink session from object 2012-04-10 18:00:26.250709 [DEBUG] switch_core_session.c:2287 sofia/sipinterface_1/7607058888@IP2 skip receive message [APPLICATION_EXEC_COMPLETE] nel is hungup already) 2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:417 (sofia/sipinterface_1/7607058888@IP2) State EXECUTE going to sleep 2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/7607058888@IP2) Running State Change CS_HANGUP 2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:602 (sofia/sipinterface_1/7607058888@IP2) State HANGUP 2012-04-10 18:00:26.250709 [DEBUG] mod_sofia.c:469 Channel sofia/sipinterface_1/7607058888@IP2 hanging up, cause: NORMAL_CLEARING 2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:47 sofia/sipinterface_1/7607058888@IP2 Standard HANGUP, cause: NORMAL_CLEARING 2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:602 (sofia/sipinterface_1/7607058888@IP2) State HANGUP going to sleep 2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:393 (sofia/sipinterface_1/7607058888@IP2) State Change CS_HANGUP -> CS_REPORTING 2012-04-10 18:00:26.250709 [DEBUG] switch_core_session.c:1182 Send signal sofia/sipinterface_1/7607058888@IP2 [BREAK] 2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:362 (sofia/sipinterface_1/7607058888@IP2) Running State Change CS_REPORTING 2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:662 (sofia/sipinterface_1/7607058888@IP2) State REPORTING 2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:79 sofia/sipinterface_1/7607058888@IP2 Standard REPORTING, cause: NORMAL_CLEARING 2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:662 (sofia/sipinterface_1/7607058888@IP2) State REPORTING going to sleep 2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:387 (sofia/sipinterface_1/7607058888@IP2) State Change CS_REPORTING -> CS_DESTROY 2012-04-10 18:00:26.250709 [DEBUG] switch_core_session.c:1182 Send signal sofia/sipinterface_1/7607058888@IP2 [BREAK] 2012-04-10 18:00:26.250709 [DEBUG] switch_core_session.c:1382 Session 1 (sofia/sipinterface_1/7607058888@IP2) Locked, Waiting on external entities 2012-04-10 18:00:26.250709 [NOTICE] switch_core_session.c:1400 Session 1 (sofia/sipinterface_1/7607058888@IP2) Ended 2012-04-10 18:00:26.250709 [NOTICE] switch_core_session.c:1402 Close Channel sofia/sipinterface_1/7607058888@IP2 [CS_DESTROY] 2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:491 (sofia/sipinterface_1/7607058888@IP2) Callstate Change HANGUP -> DOWN 2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:494 (sofia/sipinterface_1/7607058888@IP2) Running State Change CS_DESTROY 2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:504 (sofia/sipinterface_1/7607058888@IP2) State DESTROY 2012-04-10 18:00:26.250709 [DEBUG] mod_sofia.c:374 sofia/sipinterface_1/7607058888@IP2 SOFIA DESTROY 2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:86 sofia/sipinterface_1/7607058888@IP2 Standard DESTROY 2012-04-10 18:00:26.250709 [DEBUG] switch_core_state_machine.c:504 (sofia/sipinterface_1/7607058888@IP2) State DESTROY going to sleep ----- Thanks for help. Am I doing something wrong here? Malay On Tue, Apr 10, 2012 at 3:14 PM, Michael Collins <msc@freeswitch.org> wrote:

On Tue, Apr 10, 2012 at 12:22 PM, Steve Underwood <steveu@coppice.org>wrote:

...

True, but sufficient demand could change that. :-)

Steve

Good to know! Next question would be: what constitutes "sufficient demand"? :) In any case, if you are looking for g729 licenses on a platform other than Linux then let me know off list. We'll start a pool and if enough people throw their pennies in we'll move forward.

-MC

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Thanks for response but I would really like some help on why I am not able to use any other codec than PCMU/PCMA? And, if I only use WAV files from my call flow (StreamFile), do I need G729 professional license? Thanks. Malay On Tue, Apr 10, 2012 at 10:54 PM, Mitch Capper <mitch.capper@gmail.com>wrote:

I certainly would like to hear how much demand would be required for legal windows g729 support. For mobile softphone developers would be a nice option for people to be able to purchase g729 support:)

~Mitch

_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com

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Understood your point about G722. But if my list has GSM at higher priority over PCMU/PCMA and far-end SDP supports GSM then why isn't the call using GSM? I don't know where G726 comparisons are coming from. If I haven't mentioned it in vars.xml and far-end SDP doesn't support it then why would it even consider that codec? Thanks. Malay On Wed, Apr 11, 2012 at 12:53 PM, Paul Cupis <paul@cupis.co.uk> wrote:

On 11/04/12 00:12, Malay Thakershi wrote:

...

Thanks for information.

But I am not able to use any other codec than PCMU/PCMA for some reason. I read related settings and what they mean.

I am using Vitelity for now and they say they do support G722/GSM/G729 on their end.

The far-end SDP says it supports PCMU/PCMA/GSM/G729 only, no G722.

...

In vars.xml, I have <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,GSM,PCMU,PCMA"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722,GSM,PCMU,PCMA"/>

The only matching codecs are PCMA/PCMU/GSM from what you've shown us. You have some G726 comparisons in your log as well, but you've not mentioned G726 at all.

...

2012-04-10 18:00:17.430961 [DEBUG] sofia.c:5572 Remote SDP: v=0 o=root 15681 15681 IN IP4 IP2 s=session c=IN IP4 IP2 t=0 0 m=audio 12860 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

Audio Codec Compare [G729:18:8000:20:8000]/[LPC:7:8000:90:2400] Audio Codec Compare [telephone-event:101:8000:20:0]/[LPC:7:8000:90:2400] Set 2833 dtmf send/recv payload to 101 Audio Codec Compare [PCMU:0:8000:20:64000]/[G726-16:124:8000:20:16000] Audio Codec Compare [PCMA:8:8000:20:64000]/[G726-16:124:8000:20:16000] Audio Codec Compare [GSM:3:8000:20:13200]/[G726-16:124:8000:20:16000] Audio Codec Compare [G729:18:8000:20:8000]/[G726-16:124:8000:20:16000] Audio Codec Compare [telephone-event:101:8000:20:0]/[G726-16:124:8000:20:16000] Set 2833 dtmf send/recv payload to 101 Audio Codec Compare [PCMU:0:8000:20:64000]/[AAL2-G726-16:124:8000:20:16000] Audio Codec Compare [PCMA:8:8000:20:64000]/[AAL2-G726-16:124:8000:20:16000] Audio Codec Compare [GSM:3:8000:20:13200]/[AAL2-G726-16:124:8000:20:16000] Audio Codec Compare [G729:18:8000:20:8000]/[AAL2-G726-16:124:8000:20:16000] Audio Codec Compare [telephone-event:101:8000:20:0]/[AAL2-G726-16:124:8000:20:16000] Set 2833 dtmf send/recv payload to 101 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMA:8:8000:20:64000] Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] Bah HUMBUG! Sticking with PCMA@8000h@20i

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Under my sipinterface_1 profile that I use for these calls, I have: <param name="inbound-codec-negotiation" value="greedy"/> On Wed, Apr 11, 2012 at 1:43 PM, Bzzz <lazyvirus@gmx.com> wrote:

On Wed, 11 Apr 2012 13:31:45 -0500 Malay Thakershi <mthakershi@gmail.com> wrote:

...

Understood your point about G722. But if my list has GSM at higher priority over PCMU/PCMA and far-end SDP supports GSM then why isn't the call using GSM?

I don't know where G726 comparisons are coming from. If I haven't mentioned it in vars.xml and far-end SDP doesn't support it then why would it even consider that codec?

From: wiki.freeswitch.org/wiki/Sofia_Configuration_Files (inbound-codec-negotiation): 'generous' permits the remote codec list have precedence and 'win' the codec negotiation and selection process

JY -- Don't drop acid -- take it pass/fail. -- Seen in a Ladies' Room at Harvard

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1.You need OUTGOING leg to use GSM codec, right? (You wont be able to use G729 right now as it needs the license). By OUTGOING I mean call from FreeSWITCH to your provider, and not back. Yes. I want to try some codec that will use less bandwidth and see how much audio quality is compromised. So I chose GSM because G729 has license requirements.

I want this to work when someone calls my DID or my FS program calls someone via DID. So both for incoming calls as well as outgoing calls.

2. If #1 is true, then inbound-codec-negotiation is not much useful at this point. Because this setting is for INCOMING leg, and not for OUTGOING. And even if it _might_ be using GSM for INCOMING leg, but still transcode to the OUTGOING PCMA/PCMU by some reason (this is what we need to find out - why). I am not very familiar with internals of FS so having lot of trouble figuring this out.

Weird thing is, even if I do not have any mention of G729, it is being negotiated and used. Here are my configuration lines: vars.xml <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA"/> sip profile has: <param name="inbound-codec-negotiation" value="greedy"/> dial plan <extension name="CH"> <condition field="destination_number" expression="^105\d$"> <action application="set" data="absolute_codec_string=GSM"/> <action application="sleep" data="1000"/> <action application="managed" data="clsAsmtApp"/> </condition> </extension> <extension name="vitel-outbound"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <action application="set" data="absolute_codec_string=GSM"/> <action application="set" data="effective_caller_id_number=18770001111"/> <action application="bridge" data="sofia/gateway/vitelity-outbound/$1"/> </condition> </extension>

3. Try using absolute_codec_string before bridging your call to your SIP provider:

<action application="set" data="absolute_codec_string=GSM"/> <action application="bridge" data="sofia/gateway/your_provider/12345678901_example"/>

I did as you said. But found what I described above. If I remove "greedy" setting, system simply uses PCMU codec like before. This thing is so confusing. Even my SIP provider is saying they can't help out with FS because some guy quit who knew the configurations. Thanks for help. On Wed, Apr 11, 2012 at 4:12 PM, Vitalie Colosov <vetali100@gmail.com> wrote:

Lets confirm the following statements now:

1.You need OUTGOING leg to use GSM codec, right? (You wont be able to use G729 right now as it needs the license). By OUTGOING I mean call from FreeSWITCH to your provider, and not back.

2. If #1 is true, then inbound-codec-negotiation is not much useful at this point. Because this setting is for INCOMING leg, and not for OUTGOING. And even if it _might_ be using GSM for INCOMING leg, but still transcode to the OUTGOING PCMA/PCMU by some reason (this is what we need to find out - why).

3. Try using absolute_codec_string before bridging your call to your SIP provider:

<action application="set" data="absolute_codec_string=GSM"/> <action application="bridge" data="sofia/gateway/your_provider/12345678901_example"/>

And reply back if this changes anything...

Vitalie

2012/4/11 Malay Thakershi <mthakershi@gmail.com>

...

Under my sipinterface_1 profile that I use for these calls, I have:

<param name="inbound-codec-negotiation" value="greedy"/>

On Wed, Apr 11, 2012 at 1:43 PM, Bzzz <lazyvirus@gmx.com> wrote:

...

On Wed, 11 Apr 2012 13:31:45 -0500 Malay Thakershi <mthakershi@gmail.com> wrote:

...

Understood your point about G722. But if my list has GSM at higher priority over PCMU/PCMA and far-end SDP supports GSM then why isn't the call using GSM?

I don't know where G726 comparisons are coming from. If I haven't mentioned it in vars.xml and far-end SDP doesn't support it then why would it even consider that codec?

From: wiki.freeswitch.org/wiki/Sofia_Configuration_Files (inbound-codec-negotiation): 'generous' permits the remote codec list have precedence and 'win'   the codec negotiation and selection process

JY -- Don't drop acid -- take it pass/fail.                -- Seen in a Ladies' Room at Harvard

_________________________________________________________________________ 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

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Thanks. Before I do what you said, I have another problem. Sorry about that. I use different mechanism to make outgoing calls (originate calls from ESL). So I do not use bridge. I tried to put "{absolute_codec_string=GSM}" in originate call along with other parameters I pass but it throws error. My originate call looks like: originate {absolute_codec_string=GSM,ignore_early_media=true,origination_caller_id_name=CH,origination_caller_id_number=1877XXXXXXX,originate_timeout=45,chivr_outbound=true,other_var=1}sofia/gateway/vitelity-outbound/1972XXXXXXX 1056 XML default Error thrown on the log: -ERR INCOMPATIBLE_DESTINATION 2012-04-12 19:14:01.589739 [NOTICE] sofia.c:6332 Hangup sofia/sipinterface_1/1972XXXXXXX [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] Call without absolute_codec_string works fine. The way this works is: call gets originated -- it goes to default dial plan extension 1056 -- mod_managed is called from this point to handle call flow. <extension name="CH1"> <!-- Calls from public.xml will come with 1056 extension and will be handled here --> <condition field="destination_number" expression="^1056$"> <!-- codec trial setting --> <!-- <action application="set" data="absolute_codec_string=GSM"/> --> <!-- <action application="log" data="codec debug: codec_string is ${codec_string}" /> <action application="log" data="codec debug: originator_codec is ${originator_codec}" /> <action application="log" data="codec debug: absolute_codec_string is ${absolute_codec_string}" /> --> <action application="sleep" data="1000" /> <action application="managed" data="clsAsmtApp" /> </condition> </extension> As you can see, I tried setting absolute string from here too but it didn't work. Let me know what can be done if you can. Thanks. Malay On Wed, Apr 11, 2012 at 5:48 PM, Vitalie Colosov <vetali100@gmail.com> wrote:

Lets forget for a minute about vars.xml and what is there (when you use absolute_codec_string=GSM, it overrides any settings from vars.xml), and concentrate at the OUTGOING call from FreeSWITCH to your sip provider first.

Try to change your bridge to the following string: <action application="bridge"> data="{absolute_codec_string=GSM}sofia/gateway/your_provider/12345678901_example"/>

(obviously use your sip provider and phone number :)

And do one outgoing call to this sip provider.

Then, reply back whether the call was successful or not (did you hear the voice and whether the other party did hear you). And paste the log to the pastebin (http://pastebin.freeswitch.org/) and reply back with the link.

2012/4/11 Malay Thakershi <mthakershi@gmail.com>

...

...

1.You need OUTGOING leg to use GSM codec, right? (You wont be able to use G729 right now as it needs the license). By OUTGOING I mean call from FreeSWITCH to your provider, and not back. Yes. I want to try some codec that will use less bandwidth and see how much audio quality is compromised. So I chose GSM because G729 has license requirements.

I want this to work when someone calls my DID or my FS program calls someone via DID. So both for incoming calls as well as outgoing calls.

...

2. If #1 is true, then inbound-codec-negotiation is not much useful at this point. Because this setting is for INCOMING leg, and not for OUTGOING. And even if it _might_ be using GSM for INCOMING leg, but still transcode to the OUTGOING PCMA/PCMU by some reason (this is what we need to find out - why). I am not very familiar with internals of FS so having lot of trouble figuring this out.

Weird thing is, even if I do not have any mention of G729, it is being negotiated and used. Here are my configuration lines: vars.xml  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA"/>  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA"/>

sip profile has: <param name="inbound-codec-negotiation" value="greedy"/>

dial plan        <extension name="CH">      <condition field="destination_number" expression="^105\d$">                <action application="set" data="absolute_codec_string=GSM"/>                <action application="sleep" data="1000"/>                <action application="managed" data="clsAsmtApp"/>      </condition>          </extension>

 <extension name="vitel-outbound">   <condition field="destination_number" expression="^(1{0,1}\d{10})$">         <action application="set" data="absolute_codec_string=GSM"/>     <action application="set" data="effective_caller_id_number=18770001111"/>     <action application="bridge" data="sofia/gateway/vitelity-outbound/$1"/>    </condition>  </extension>

...

3. Try using absolute_codec_string before bridging your call to your SIP provider:

<action application="set" data="absolute_codec_string=GSM"/> <action application="bridge" data="sofia/gateway/your_provider/12345678901_example"/>

I did as you said. But found what I described above. If I remove "greedy" setting, system simply uses PCMU codec like before.

This thing is so confusing. Even my SIP provider is saying they can't help out with FS because some guy quit who knew the configurations.

Thanks for help.

On Wed, Apr 11, 2012 at 4:12 PM, Vitalie Colosov <vetali100@gmail.com> wrote:

...

Lets confirm the following statements now:

1.You need OUTGOING leg to use GSM codec, right? (You wont be able to use G729 right now as it needs the license). By OUTGOING I mean call from FreeSWITCH to your provider, and not back.

2. If #1 is true, then inbound-codec-negotiation is not much useful at this point. Because this setting is for INCOMING leg, and not for OUTGOING. And even if it _might_ be using GSM for INCOMING leg, but still transcode to the OUTGOING PCMA/PCMU by some reason (this is what we need to find out - why).

3. Try using absolute_codec_string before bridging your call to your SIP provider:

<action application="set" data="absolute_codec_string=GSM"/> <action application="bridge" data="sofia/gateway/your_provider/12345678901_example"/>

And reply back if this changes anything...

Vitalie

2012/4/11 Malay Thakershi <mthakershi@gmail.com>

...

Under my sipinterface_1 profile that I use for these calls, I have:

<param name="inbound-codec-negotiation" value="greedy"/>

On Wed, Apr 11, 2012 at 1:43 PM, Bzzz <lazyvirus@gmx.com> wrote:

...

On Wed, 11 Apr 2012 13:31:45 -0500 Malay Thakershi <mthakershi@gmail.com> wrote:

...

Understood your point about G722. But if my list has GSM at higher priority over PCMU/PCMA and far-end SDP supports GSM then why isn't the call using GSM?

I don't know where G726 comparisons are coming from. If I haven't mentioned it in vars.xml and far-end SDP doesn't support it then why would it even consider that codec?

From: wiki.freeswitch.org/wiki/Sofia_Configuration_Files (inbound-codec-negotiation): 'generous' permits the remote codec list have precedence and 'win'   the codec negotiation and selection process

JY -- Don't drop acid -- take it pass/fail.                -- Seen in a Ladies' Room at Harvard

_________________________________________________________________________ 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

_________________________________________________________________________ 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

_________________________________________________________________________ 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

When I do "sofia global siptrace on", my console just doesn't stop. I get text so fast, I can't even paste the originate command. So I can't produce what will hep you help me on this. Is there a way I can divert the whole thing to a text file? Thanks. Malay On Thu, Apr 12, 2012 at 10:55 PM, Vitalie Colosov <vetali100@gmail.com> wrote:

Could you collect the full siptrace of one call and paste into pastebin?

To do so, type at the cli:

       sofia global siptrace on

Then make 1 call and collect whatever you got at the screen, and paste into pastebin.

Then, you'll probably want to disable siptrace:

        sofia global siptrace off

Regards, Vitalie

2012/4/12 Malay Thakershi <mthakershi@gmail.com>

...

Thanks.

Before I do what you said, I have another problem. Sorry about that.

I use different mechanism to make outgoing calls (originate calls from ESL). So I do not use bridge. I tried to put "{absolute_codec_string=GSM}" in originate call along with other parameters I pass but it throws error.

My originate call looks like: originate {absolute_codec_string=GSM,ignore_early_media=true,origination_caller_id_name=CH,origination_caller_id_number=1877XXXXXXX,originate_timeout=45,chivr_outbound=true,other_var=1}sofia/gateway/vitelity-outbound/1972XXXXXXX 1056 XML default

Error thrown on the log: -ERR INCOMPATIBLE_DESTINATION 2012-04-12 19:14:01.589739 [NOTICE] sofia.c:6332 Hangup sofia/sipinterface_1/1972XXXXXXX [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]

Call without absolute_codec_string works fine.

The way this works is: call gets originated -- it goes to default dial plan extension 1056 -- mod_managed is called from this point to handle call flow.

   <extension name="CH1">          <!-- Calls from public.xml will come with 1056 extension and will be handled here -->      <condition field="destination_number" expression="^1056$">        <!-- codec trial setting -->        <!-- <action application="set" data="absolute_codec_string=GSM"/> --> <!--            <action application="log" data="codec debug: codec_string is ${codec_string}" />                <action application="log" data="codec debug: originator_codec is ${originator_codec}" />                <action application="log" data="codec debug: absolute_codec_string is ${absolute_codec_string}" />  -->                <action application="sleep" data="1000" />        <action application="managed" data="clsAsmtApp" />      </condition>    </extension>

As you can see, I tried setting absolute string from here too but it didn't work.

Let me know what can be done if you can.

Thanks.

Malay

On Wed, Apr 11, 2012 at 5:48 PM, Vitalie Colosov <vetali100@gmail.com> wrote:

...

Lets forget for a minute about vars.xml and what is there (when you use absolute_codec_string=GSM, it overrides any settings from vars.xml), and concentrate at the OUTGOING call from FreeSWITCH to your sip provider first.

Try to change your bridge to the following string: <action application="bridge">

data="{absolute_codec_string=GSM}sofia/gateway/your_provider/12345678901_example"/>

(obviously use your sip provider and phone number :)

And do one outgoing call to this sip provider.

Then, reply back whether the call was successful or not (did you hear the voice and whether the other party did hear you). And paste the log to the pastebin (http://pastebin.freeswitch.org/) and reply back with the link.

2012/4/11 Malay Thakershi <mthakershi@gmail.com>

...

...

1.You need OUTGOING leg to use GSM codec, right? (You wont be able to use G729 right now as it needs the license). By OUTGOING I mean call from FreeSWITCH to your provider, and not back. Yes. I want to try some codec that will use less bandwidth and see how much audio quality is compromised. So I chose GSM because G729 has license requirements.

I want this to work when someone calls my DID or my FS program calls someone via DID. So both for incoming calls as well as outgoing calls.

...

2. If #1 is true, then inbound-codec-negotiation is not much useful at this point. Because this setting is for INCOMING leg, and not for OUTGOING. And even if it _might_ be using GSM for INCOMING leg, but still transcode to the OUTGOING PCMA/PCMU by some reason (this is what we need to find out - why). I am not very familiar with internals of FS so having lot of trouble figuring this out.

Weird thing is, even if I do not have any mention of G729, it is being negotiated and used. Here are my configuration lines: vars.xml  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM,PCMU,PCMA"/>  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA"/>

sip profile has: <param name="inbound-codec-negotiation" value="greedy"/>

dial plan        <extension name="CH">      <condition field="destination_number" expression="^105\d$">                <action application="set" data="absolute_codec_string=GSM"/>                <action application="sleep" data="1000"/>                <action application="managed" data="clsAsmtApp"/>      </condition>          </extension>

 <extension name="vitel-outbound">   <condition field="destination_number" expression="^(1{0,1}\d{10})$">         <action application="set" data="absolute_codec_string=GSM"/>     <action application="set" data="effective_caller_id_number=18770001111"/>     <action application="bridge" data="sofia/gateway/vitelity-outbound/$1"/>    </condition>  </extension>

...

3. Try using absolute_codec_string before bridging your call to your SIP provider:

<action application="set" data="absolute_codec_string=GSM"/> <action application="bridge" data="sofia/gateway/your_provider/12345678901_example"/>

I did as you said. But found what I described above. If I remove "greedy" setting, system simply uses PCMU codec like before.

This thing is so confusing. Even my SIP provider is saying they can't help out with FS because some guy quit who knew the configurations.

Thanks for help.

On Wed, Apr 11, 2012 at 4:12 PM, Vitalie Colosov <vetali100@gmail.com> wrote:

...

Lets confirm the following statements now:

1.You need OUTGOING leg to use GSM codec, right? (You wont be able to use G729 right now as it needs the license). By OUTGOING I mean call from FreeSWITCH to your provider, and not back.

2. If #1 is true, then inbound-codec-negotiation is not much useful at this point. Because this setting is for INCOMING leg, and not for OUTGOING. And even if it _might_ be using GSM for INCOMING leg, but still transcode to the OUTGOING PCMA/PCMU by some reason (this is what we need to find out - why).

3. Try using absolute_codec_string before bridging your call to your SIP provider:

<action application="set" data="absolute_codec_string=GSM"/> <action application="bridge" data="sofia/gateway/your_provider/12345678901_example"/>

And reply back if this changes anything...

Vitalie

2012/4/11 Malay Thakershi <mthakershi@gmail.com>

...

Under my sipinterface_1 profile that I use for these calls, I have:

<param name="inbound-codec-negotiation" value="greedy"/>

On Wed, Apr 11, 2012 at 1:43 PM, Bzzz <lazyvirus@gmx.com> wrote: > > On Wed, 11 Apr 2012 13:31:45 -0500 > Malay Thakershi <mthakershi@gmail.com> wrote: > > > Understood your point about G722. But if my list has GSM at > > higher > > priority > > over PCMU/PCMA and far-end SDP supports GSM then why isn't the > > call > > using > > GSM? > > > > I don't know where G726 comparisons are coming from. If I haven't > > mentioned > > it in vars.xml and far-end SDP doesn't support it then why would > > it > > even > > consider that codec? > > From: wiki.freeswitch.org/wiki/Sofia_Configuration_Files > (inbound-codec-negotiation): > 'generous' permits the remote codec list have precedence and 'win' >   the codec negotiation and selection process > > JY > -- > Don't drop acid -- take it pass/fail. >                -- Seen in a Ladies' Room at Harvard > > > > > _________________________________________________________________________ > 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

_________________________________________________________________________ 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

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_________________________________________________________________________ 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

_________________________________________________________________________ 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