[Freeswitch-users] Possible to disable core codecs?
Malay Thakershi
mthakershi at gmail.com
Tue Apr 10 09:22:47 MSD 2012
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 at 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 at 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 at 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 at 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 at IP1 SOFI
A INIT
2012-04-10 00:13:32.960773 [DEBUG] mod_sofia.c:125
(sofia/sipinterface_1/Phone2 at 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 at 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 at 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 at IP1 SOF
IA ROUTING
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:104
sofia/sipinterface_1/Phone2 at 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 at IP1 parsing [inbound->vitel-inbound]
continue=fa
lse
Dialplan: sofia/sipinterface_1/Phone2 at IP1 Regex (PASS) [vitel-inbound]
destination_num
ber(8774542559) =~ // break=on-false
Dialplan: sofia/sipinterface_1/Phone2 at 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 at 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 at IP1 SOF
IA EXECUTE
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:192
sofia/sipinterface_1/Phone2 at 6
6.241.99.201 Standard EXECUTE
EXECUTE sofia/sipinterface_1/Phone2 at IP1 transfer(1056 XML default)
2012-04-10 00:13:32.960773 [DEBUG] switch_ivr.c:1711
(sofia/sipinterface_1/Phone2 at 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 at IP1 [BREAK]
2012-04-10 00:13:32.960773 [DEBUG] switch_core_session.c:731 Send signal
sofia/sipinterface_1/972782
9132 at IP1 [BREAK]
2012-04-10 00:13:32.960773 [NOTICE] switch_ivr.c:1717 Transfer
sofia/sipinterface_1/Phone2 at x.24
1.99.201 to XML[1056 at 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 at IP1 SOF
IA ROUTING
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:104
sofia/sipinterface_1/Phone2 at 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 at IP1 parsing [default->CHPhoneAsmtDev]
continue=f
alse
Dialplan: sofia/sipinterface_1/Phone2 at IP1 Regex (PASS) [CHPhoneAsmtDev]
destination_nu
mber(1056) =~ /^105\d$/ break=on-false
Dialplan: sofia/sipinterface_1/Phone2 at IP1 Action sleep(1000)
Dialplan: sofia/sipinterface_1/Phone2 at 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 at 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 at IP1 SOF
IA EXECUTE
2012-04-10 00:13:32.960773 [DEBUG] switch_core_state_machine.c:192
sofia/sipinterface_1/Phone2 at 6
6.241.99.201 Standard EXECUTE
EXECUTE sofia/sipinterface_1/Phone2 at IP1 sleep(1000)
EXECUTE sofia/sipinterface_1/Phone2 at 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 at 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 at 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 at IP1 [BREAK]
2012-04-10 00:13:34.780969 [DEBUG] switch_channel.c:3244
(sofia/sipinterface_1/Phone2 at 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 at 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 at IP1 [BREAK]
2012-04-10 00:13:34.780969 [DEBUG] sofia.c:5561 Channel
sofia/sipinterface_1/Phone2 at 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 at IP1 [BREAK]
2012-04-10 00:13:34.821006 [DEBUG] switch_core_session.c:877 Send signal
sofia/sipinterface_1/972782
9132 at IP1 [BREAK]
2012-04-10 00:13:34.821006 [DEBUG] switch_core_session.c:877 Send signal
sofia/sipinterface_1/972782
9132 at 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 at 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 at 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 at avimarcus.net> wrote:
> Try absolute_codec_string<http://wiki.freeswitch.org/wiki/Variable_absolute_codec_string>
> -Avi
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
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