[Freeswitch-users] Audio files are not bein played

[Vikas Kumar]

Hi All,
We are using Linphone and zoiper on other hand.   We have some limit in dial plan that only one call is admissible at one time.
If I call from other softphone while call is in progress  than this limit is hit. See logs:


Dialplan: sofia/internal/1016 at 119.9.73.183<mailto:sofia/internal/1016 at 119.9.73.183> Action playback(/usr/local/videocallhandler/media/InQue1.wav)
Dialplan: sofia/internal/1016 at 119.9.73.183<mailto:sofia/internal/1016 at 119.9.73.183> Action playback(/usr/local/videocallhandler/media/id.wav)
Dialplan: sofia/internal/1016 at 119.9.73.183<mailto:sofia/internal/1016 at 119.9.73.183> Action playback(/usr/local/videocallhandler/media/InQue2.wav)
Dialplan: sofia/internal/1016 at 119.9.73.183<mailto:sofia/internal/1016 at 119.9.73.183> Action endless_playback(/tmp/times.wav)
2015-09-22 11:38:43.542214 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/1016 at 119.9.73.183<mailto:sofia/internal/1016 at 119.9.73.183>) State Change CS_ROUTING -> CS_EXECUTE
2015-09-22 11:38:43.542214 [DEBUG] switch_core_session.c:1357 Send signal sofia/internal/1016 at 119.9.73.183<mailto:sofia/internal/1016 at 119.9.73.183> [BREAK]
2015-09-22 11:38:43.542214 [DEBUG] switch_core_state_machine.c:474 (sofia/internal/1016 at 119.9.73.183<mailto:sofia/internal/1016 at 119.9.73.183>) State ROUTING going to sleep
2015-09-22 11:38:43.542214 [DEBUG] switch_core_state_machine.c:418 (sofia/internal/1016 at 119.9.73.183<mailto:sofia/internal/1016 at 119.9.73.183>) Running State Change CS_EXECUTE
2015-09-22 11:38:43.542214 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/1016 at 119.9.73.183<mailto:sofia/internal/1016 at 119.9.73.183>) State EXECUTE
2015-09-22 11:38:43.542214 [DEBUG] mod_sofia.c:243 sofia/internal/1016 at 119.9.73.183<mailto:sofia/internal/1016 at 119.9.73.183> SOFIA EXECUTE
2015-09-22 11:38:43.542214 [DEBUG] switch_core_state_machine.c:209 sofia/internal/1016 at 119.9.73.183<mailto:sofia/internal/1016 at 119.9.73.183> Standard EXECUTE
EXECUTE sofia/internal/1016 at 119.9.73.183<mailto:sofia/internal/1016 at 119.9.73.183> answer()
2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:5308 Audio Codec Compare [GSM:3:8000:20:13200]/[G729:18:8000:20:8000]
2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:5308 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000]
2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:5308 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000]
2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:5308 Audio Codec Compare [GSM:3:8000:20:13200]/[GSM:3:8000:20:13200]
2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:3216 Set Codec sofia/internal/1016 at 119.9.73.183<mailto:sofia/internal/1016 at 119.9.73.183> GSM/8000 20 ms 160 samples 13200 bits
2015-09-22 11:38:43.542214 [DEBUG] switch_core_codec.c:111 sofia/internal/1016 at 119.9.73.183<mailto:sofia/internal/1016 at 119.9.73.183> Original read codec set to GSM:3
2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:5477 Set 2833 dtmf send/recv payload to 101
2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:3475 AUDIO RTP [sofia/internal/1016 at 119.9.73.183] 119.9.73.183 port 25094 -> 10.150.178.79 port 60204 codec: 3 ms: 20
2015-09-22 11:38:43.542214 [DEBUG] switch_rtp.c:2041 Starting timer [soft] 160 bytes per 20ms
2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:3742 Set 2833 dtmf send payload to 101
2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:3748 Set 2833 dtmf receive payload to 101
2015-09-22 11:38:43.542214 [DEBUG] sofia_glue.c:3775 sofia/internal/1016 at 119.9.73.183<mailto:sofia/internal/1016 at 119.9.73.183> Set rtp dtmf delay to 40
2015-09-22 11:38:43.542214 [NOTICE] sofia_glue.c:4386 Pre-Answer sofia/internal/1016 at 119.9.73.183<mailto:sofia/internal/1016 at 119.9.73.183>!
2015-09-22 11:38:43.542214 [DEBUG] switch_channel.c:3370 (sofia/internal/1016 at 119.9.73.183<mailto:sofia/internal/1016 at 119.9.73.183>) Callstate Change RINGING -> EARLY
2015-09-22 11:38:43.542214 [DEBUG] mod_sofia.c:866 Local SDP sofia/internal/1016 at 119.9.73.183<mailto:sofia/internal/1016 at 119.9.73.183>:
v=0
o=FreeSWITCH 1442877029 1442877030 IN IP4 119.9.73.183
s=FreeSWITCH
c=IN IP4 119.9.73.183
t=0 0
m=audio 25094 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv



Warning given is "/usr/local/videocallhandler/media/Hold.wav sample rate 11025 doesn't match requested rate 8000"

Should I ignore this warning, ??

Thanks
Vikas


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