Any solution to this. Any help will be appreciated.

Thanks

Vikas

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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