Hi Team, I made a script to dial a user , and if not answered the second user should get the dial. Everything works file , Except the first user is dialed for 3 times , if the user is busy or unanswered , i was trying to figure out what could be the problem for this issue, but i couldn't . Can Anyone Figure out what is the reason for this issue. below is my lua dial script session:execute("bridge","{ignore_early_media=true,monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7,originate_continue_on_timeout=true}sofia/gateway/bandwidth.com/+919xxxxxxxx|sofia/gateway/bandwidth.com/+919xxxxxxxx"); following is the log i can get. Any Help or suggestion would be really helpful! [NOTICE] sofia.c:5594 Ring-Ready sofia/internal/+919677080275! 2012-08-01 07:09:13.212635 [DEBUG] switch_ivr_originate.c:1164 Raw Codec Activation Success L16@16000hz 1 channel 20ms 2012-08-01 07:09:13.212635 [DEBUG] switch_core_codec.c:216 rtmp/default/+542914850488 Push codec L16:70 2012-08-01 07:09:13.212635 [DEBUG] switch_ivr_originate.c:1227 Play Ringback Tone [%(2000,4000,440,480)] 2012-08-01 07:09:16.532655 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/+919677080275 [BREAK] 2012-08-01 07:09:16.532655 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/+919677080275 [BREAK] 2012-08-01 07:09:16.552637 [DEBUG] sofia.c:5502 Channel sofia/internal/+919677080275 entering state [proceeding][183] 2012-08-01 07:09:16.552637 [DEBUG] sofia.c:5513 Remote SDP: v=0 o=Sonus_UAC 31326 20180 IN IP4 65.115.130.14 s=SIP Media Capabilities c=IN IP4 65.115.130.14 t=0 0 m=audio 10382 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[SPEEX:99:16000:20:42200] 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000] 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000] 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:2919 Set Codec sofia/internal/+919677080275 PCMU/8000 20 ms 160 samples 64000 bits 2012-08-01 07:09:16.552637 [DEBUG] switch_core_codec.c:111 sofia/internal/+919677080275 Original read codec set to PCMU:0 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send payload to 101 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:3171 AUDIO RTP [sofia/internal/+919677080275] 10.244.15.45 port 30506 -> 65.115.130.14 port 10382 codec: 0 ms: 20 2012-08-01 07:09:16.552637 [DEBUG] switch_rtp.c:1661 Starting timer [soft] 160 bytes per 20ms 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:3435 Set 2833 dtmf send payload to 101 2012-08-01 07:09:16.552637 [DEBUG] sofia_glue.c:3441 Set 2833 dtmf receive payload to 101 2012-08-01 07:09:16.552637 [NOTICE] sofia_glue.c:3945 Pre-Answer sofia/internal/+919677080275! 2012-08-01 07:09:16.552637 [DEBUG] switch_channel.c:2936 (sofia/internal/+919677080275) Callstate Change RINGING -> EARLY 2012-08-01 07:09:16.552637 [DEBUG] switch_channel.c:2978 Send signal rtmp/default/+542914850488 [BREAK] 2012-08-01 07:09:16.572648 [DEBUG] switch_ivr_async.c:2782 Adding tone spec 480,620 index 0 hits 1 2012-08-01 07:09:16.572648 [DEBUG] switch_core_media_bug.c:457 Attaching BUG to sofia/internal/+919677080275 2012-08-01 07:09:16.572648 [DEBUG] switch_ivr_async.c:2782 Adding tone spec 1776.7 index 1 hits 2 2012-08-01 07:09:16.572648 [DEBUG] switch_ivr_async.c:2877 sofia/internal/+919677080275 bug already running 2012-08-01 07:09:16.572648 [DEBUG] switch_rtp.c:3205 Correct ip/port confirmed. 2012-08-01 07:09:16.572648 [DEBUG] switch_core_io.c:340 Setting BUG Codec PCMU:0 2012-08-01 07:09:33.072650 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/+919677080275 [BREAK] 2012-08-01 07:09:33.072650 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/+919677080275 [BREAK] 2012-08-01 07:09:33.092641 [DEBUG] sofia.c:5502 Channel sofia/internal/+919677080275 entering state [proceeding][183] 2012-08-01 07:09:33.092641 [DEBUG] sofia.c:5510 Duplicate SDP v=0 o=Sonus_UAC 31326 20180 IN IP4 65.115.130.14 s=SIP Media Capabilities c=IN IP4 65.115.130.14 t=0 0 m=audio 10382 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2012-08-01 07:09:33.092641 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-08-01 07:09:33.092641 [DEBUG] sofia_glue.c:2853 Already using PCMU 2012-08-01 07:09:33.092641 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send payload to 101 2012-08-01 07:09:33.092641 [DEBUG] sofia_glue.c:3141 Audio params are unchanged for sofia/internal/+919677080275. 2012-08-01 07:09:33.092641 [DEBUG] sofia_glue.c:3151 sofia/internal/+919677080275 Setting audio receive payload in Re-INVITE to 0 2012-08-01 07:09:36.372634 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/+919677080275 [BREAK] 2012-08-01 07:09:36.372634 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/+919677080275 [BREAK] 2012-08-01 07:09:36.392686 [DEBUG] sofia.c:5502 Channel sofia/internal/+919677080275 entering state [proceeding][183] 2012-08-01 07:09:36.392686 [DEBUG] sofia.c:5510 Duplicate SDP v=0 o=Sonus_UAC 31326 20180 IN IP4 65.115.130.14 s=SIP Media Capabilities c=IN IP4 65.115.130.14 t=0 0 m=audio 10382 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2012-08-01 07:09:36.392686 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-08-01 07:09:36.392686 [DEBUG] sofia_glue.c:2853 Already using PCMU 2012-08-01 07:09:36.392686 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send payload to 101 2012-08-01 07:09:36.392686 [DEBUG] sofia_glue.c:3141 Audio params are unchanged for sofia/internal/+919677080275. 2012-08-01 07:09:36.392686 [DEBUG] sofia_glue.c:3151 sofia/internal/+919677080275 Setting audio receive payload in Re-INVITE to 0 2012-08-01 07:09:38.512638 [DEBUG] libdingaling.c:1624 Sent keep alive signal 2012-08-01 07:09:55.272635 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/+919677080275 [BREAK] 2012-08-01 07:09:55.272635 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/+919677080275 [BREAK] 2012-08-01 07:09:55.272635 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/+919677080275 [BREAK] 2012-08-01 07:09:55.272635 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/+919677080275 [BREAK] 2012-08-01 07:09:55.292634 [DEBUG] sofia.c:5502 Channel sofia/internal/+919677080275 entering state [proceeding][183] 2012-08-01 07:09:55.292634 [DEBUG] sofia.c:5510 Duplicate SDP v=0 o=Sonus_UAC 31326 20180 IN IP4 65.115.130.14 s=SIP Media Capabilities c=IN IP4 65.115.130.14 t=0 0 m=audio 10382 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:2853 Already using PCMU 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send payload to 101 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:3141 Audio params are unchanged for sofia/internal/+919677080275. 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:3151 sofia/internal/+919677080275 Setting audio receive payload in Re-INVITE to 0 2012-08-01 07:09:55.292634 [DEBUG] sofia.c:5502 Channel sofia/internal/+919677080275 entering state [proceeding][183] 2012-08-01 07:09:55.292634 [DEBUG] sofia.c:5510 Duplicate SDP v=0 o=Sonus_UAC 31326 20180 IN IP4 65.115.130.14 s=SIP Media Capabilities c=IN IP4 65.115.130.14 t=0 0 m=audio 10382 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:2853 Already using PCMU 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send payload to 101 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:3141 Audio params are unchanged for sofia/internal/+919677080275. 2012-08-01 07:09:55.292634 [DEBUG] sofia_glue.c:3151 sofia/internal/+919677080275 Setting audio receive payload in Re-INVITE to 0 2012-08-01 07:10:11.012635 [DEBUG] switch_core_codec.c:241 rtmp/default/+542914850488 Restore previous codec SPEEX:99. 2012-08-01 07:10:11.012635 [DEBUG] switch_channel.c:2852 (sofia/internal/+919677080275) Callstate Change EARLY -> HANGUP 2012-08-01 07:10:11.012635 [NOTICE] switch_ivr_originate.c:3182 Hangup sofia/internal/+919677080275 [CS_CONSUME_MEDIA] [NO_ANSWER] Thanks Ramesh -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-Auto-Retrying-a-cal... 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