[Freeswitch-users] Freeswitch Auto Retrying a call for 3 times , if it is not answered
ramesh
ramesh_mind at yahoo.com
Wed Aug 1 13:15:53 MSD 2012
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 at 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
--
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