Freeswitch Auto Retrying a call for 3 times , if it is not answered
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... Sent from the freeswitch-users mailing list archive at Nabble.com.
What does it mean that the first user is dialed "three times?" Also, put this log on pastebin.freeswitch.org and use FreeSWITCH Log as the syntax highlighting. That makes it much easier to read. Thanks, MC On Wed, Aug 1, 2012 at 2:15 AM, ramesh <ramesh_mind@yahoo.com> wrote:
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... Sent from the freeswitch-users mailing list archive at Nabble.com.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
participants (2)
-
Michael Collins
-
ramesh