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-call-for-3-times-if-it-is-not-answered-tp7581406.html
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