[Freeswitch-users] Freeswitch Auto Retrying a call for 3 times , if it is not answered
Michael Collins
msc at freeswitch.org
Thu Aug 2 20:27:40 MSD 2012
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 at 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 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
>
>
>
>
> --
> 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.
>
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>
>
>
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--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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