Hi Vikas, Probably is the the best way to convert the audio file to sample rate 8000 The following command should work (converts to mono, 8000 rate): sox hold.wav -r 8000 -c 1 hold_8000.wav rm hold.wav mv hold_8000.wav hold.wav Raphael
On 23 Sep 2015, at 06:36, Vikas Kumar <Vikas.Kumar1@timesinternet.in> wrote:
Any solution to this. Any help will be appreciated.
Thanks Vikas
From: freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Vikas Kumar Sent: 22 September 2015 21:35 To: freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] Audio files are not bein played
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@119.9.73.183 <mailto:sofia/internal/1016@119.9.73.183> Action playback(/usr/local/videocallhandler/media/InQue1.wav) Dialplan: sofia/internal/1016@119.9.73.183 <mailto:sofia/internal/1016@119.9.73.183> Action playback(/usr/local/videocallhandler/media/id.wav) Dialplan: sofia/internal/1016@119.9.73.183 <mailto:sofia/internal/1016@119.9.73.183> Action playback(/usr/local/videocallhandler/media/InQue2.wav) Dialplan: sofia/internal/1016@119.9.73.183 <mailto:sofia/internal/1016@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@119.9.73.183 <mailto:sofia/internal/1016@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@119.9.73.183 <mailto:sofia/internal/1016@119.9.73.183> [BREAK] 2015-09-22 11:38:43.542214 [DEBUG] switch_core_state_machine.c:474 (sofia/internal/1016@119.9.73.183 <mailto:sofia/internal/1016@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@119.9.73.183 <mailto:sofia/internal/1016@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@119.9.73.183 <mailto:sofia/internal/1016@119.9.73.183>) State EXECUTE 2015-09-22 11:38:43.542214 [DEBUG] mod_sofia.c:243 sofia/internal/1016@119.9.73.183 <mailto:sofia/internal/1016@119.9.73.183> SOFIA EXECUTE 2015-09-22 11:38:43.542214 [DEBUG] switch_core_state_machine.c:209 sofia/internal/1016@119.9.73.183 <mailto:sofia/internal/1016@119.9.73.183> Standard EXECUTE EXECUTE sofia/internal/1016@119.9.73.183 <mailto:sofia/internal/1016@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@119.9.73.183 <mailto:sofia/internal/1016@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@119.9.73.183 <mailto:sofia/internal/1016@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@119.9.73.183 <mailto:sofia/internal/1016@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@119.9.73.183 <mailto:sofia/internal/1016@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@119.9.73.183 <mailto:sofia/internal/1016@119.9.73.183>! 2015-09-22 11:38:43.542214 [DEBUG] switch_channel.c:3370 (sofia/internal/1016@119.9.73.183 <mailto:sofia/internal/1016@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@119.9.73.183 <mailto:sofia/internal/1016@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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