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 Action playback(/usr/local/videocallhandler/media/InQue1.wav)

Dialplan: sofia/internal/1016@119.9.73.183 Action playback(/usr/local/videocallhandler/media/id.wav)

Dialplan: sofia/internal/1016@119.9.73.183 Action playback(/usr/local/videocallhandler/media/InQue2.wav)

Dialplan: 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) 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 [BREAK]

2015-09-22 11:38:43.542214 [DEBUG] switch_core_state_machine.c:474 (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) 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) State EXECUTE

2015-09-22 11:38:43.542214 [DEBUG] mod_sofia.c:243 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 Standard EXECUTE

EXECUTE 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 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 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] 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 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!

2015-09-22 11:38:43.542214 [DEBUG] switch_channel.c:3370 (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:

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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