June 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Error while trying to install freeswitch latest source from git
by ramesh 28 Jun '12

28 Jun '12

Hi All, We are to trying to install latest freeswitch source into our new server , while trying to make the source , we are getting error and unable to proceed futhur . Any Idea? to make it install? Thanks, Ramesh -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-while-trying-to-install… Sent from the freeswitch-users mailing list archive at Nabble.com.

3 4

(no subject)
by Rodney 28 Jun '12

28 Jun '12

http://feiyankj.com/test.php?meet207.png

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 72, Issue 253
by Jose Miguel Sucasas Mejuto 28 Jun '12

28 Jun '12

Hello Peter, As you can see in the 4th line of my dialplan, I have this variable setted, but have no effect, and pauses jitterbuffer: > <action application="cng_plc"/> >  > <action application="set" data="jitterbuffer_msec=90"/> > <action application="set" data="sip_jitter_buffer_during_bridge=true"/> >  > <action application="jitterbuffer" data="debug:${uuid}"/> >  > <action application="bridge" data="{jitterbuffer_msec=90,absolute_codec_string='SILK@16000h,iLBC@30i,G729,AMR,AMR-WB,H263,H264'}user/${dialed_extension}@${domain_name}"/> Following FreeSwitch documentation, I dont understand why jitterbuffer is paused. Thanks for your reply, > De: Peter Olsson <peter.olsson(a)visionutveckling.se> > Asunto: Re: [Freeswitch-users] Always PAUSE Jitterbuffer > Fecha: 27 de junio de 2012 22:17:11 GMT+02:00 > Para: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Responder a: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > Maybe this is what you are looking for? I think FS usually wants the endpoints to handle this during a bridge, setting this var should force it to stay enabled. > > http://wiki.freeswitch.org/wiki/Variable_sip_jitter_buffer_during_bridge > > /Peter > > 27 jun 2012 kl. 22:11 skrev "Jose Miguel Sucasas Mejuto" <jmsucasas(a)me.com<mailto:jmsucasas@me.com>>: > > Hello, > > I have installed Freeswitch from GIT (currenty I have installed 1.2.0-rc2+git~20120625T203158Z~7dc9a9cacc version). Because some users connects from 3G network using a smartphone, I try to activate JitterBuffer and Packet Loss Concealing in both directions (in a call with U1 and U2, I try to activate 90ms jitterbuffer in audio flow U1->U2 and another 90ms jitterbuffer in audio flow U2->U1). To achieve this, I have this in dialplan (default.xml file): > > <action application="cng_plc"/> >  > <action application="set" data="jitterbuffer_msec=90"/> > <action application="set" data="sip_jitter_buffer_during_bridge=true"/> >  > <action application="jitterbuffer" data="debug:${uuid}"/> >  > <action application="bridge" data="{jitterbuffer_msec=90,absolute_codec_string='SILK@16000h,iLBC@30i,G729,AMR,AMR-WB,H263,H264'}user/${dialed_extension}@${domain_name}"/> > > User 1002 is on the same network as Freeswitch box, connected by ethernet line, and user 1001 is on 3G, when user 1002 calls to user 1001 (both of them uses iLBC30 codec), on fs_cli it seems that JitterBuffer is activated (in bold green) > > 2012-06-27 20:55:14.929462 [NOTICE] sofia.c:5930 Ring-Ready sofia/internal/sip:1001@10.10.57.82:50203! > 2012-06-27 20:55:14.929462 [INFO] switch_ivr_originate.c:1156 Sending early media > 2012-06-27 20:55:14.929462 [INFO] switch_nat.c:590 NAT port mapping disabled > 2012-06-27 20:55:14.929462 [INFO] switch_nat.c:590 NAT port mapping disabled > 2012-06-27 20:55:14.929462 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001@10.10.57.82:50203 entering state [proceeding][180] > 2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3269 AUDIO RTP [sofia/internal/1002@sukmad.dyndns-home.com:58180] 79.150.184.57 port 11154 -> 192.168.0.15 port 10000 codec: 98 ms: 30 > 2012-06-27 20:55:14.929462 [DEBUG] switch_rtp.c:1861 Starting timer [soft] 240 bytes per 30ms > 2012-06-27 20:55:14.929462 [DEBUG] switch_rtp.c:2207 RTCP send rate is: 5000 and packet rate is: 30000 Remote Port: 10001 > 2012-06-27 20:55:14.929462 [DEBUG] switch_rtp.c:1230 Setting RTCP remote addr to 192.168.0.15:10001 > 2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3491 Setting Jitterbuffer to 90ms (3 frames) > 2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3533 Set 2833 dtmf send payload to 101 > 2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3539 Set 2833 dtmf receive payload to 101 > 2012-06-27 20:55:14.929462 [DEBUG] sofia_glue.c:3566 sofia/internal/1002@sukmad.dyndns-home.com:58180 Set rtp dtmf delay to 40 > 2012-06-27 20:55:14.929462 [DEBUG] mod_sofia.c:2606 Ring SDP: > v=0 > o=FreeSWITCH 1340812160 1340812161 IN IP4 79.150.184.57 > s=FreeSWITCH > c=IN IP4 79.150.184.57 > t=0 0 > m=audio 11154 RTP/AVP 98 101 > a=rtpmap:98 iLBC/8000 > a=fmtp:98 mode=30 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:30 > a=sendrecv > > 2012-06-27 20:55:14.929462 [NOTICE] mod_sofia.c:2609 Pre-Answer sofia/internal/1002@sukmad.dyndns-home.com:58180! > 2012-06-27 20:55:14.929462 [DEBUG] switch_channel.c:3042 (sofia/internal/1002@sukmad.dyndns-home.com:58180) Callstate Change RINGING -> EARLY > 2012-06-27 20:55:14.929462 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/1002@sukmad.dyndns-home.com:58180 [BREAK] > 2012-06-27 20:55:14.929462 [DEBUG] switch_ivr_originate.c:1205 Raw Codec Activation Success L16@8000hz 1 channel 30ms > 2012-06-27 20:55:14.929462 [DEBUG] switch_core_codec.c:219 sofia/internal/1002@sukmad.dyndns-home.com:58180 Push codec L16:70 > 2012-06-27 20:55:14.929462 [DEBUG] switch_ivr_originate.c:1268 Play Ringback Tone [%(2000,4000,440,480)] > 2012-06-27 20:55:14.929462 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002@sukmad.dyndns-home.com:58180 [BREAK] > 2012-06-27 20:55:14.929462 [DEBUG] sofia.c:5831 Channel sofia/internal/1002@sukmad.dyndns-home.com:58180 skipping state [early][183] > 2012-06-27 20:55:14.939460 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001@31.4.238.246:22268 [BREAK] > 2012-06-27 20:55:14.939460 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001@31.4.238.246:22268 [BREAK] > 2012-06-27 20:55:14.939460 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001@31.4.238.246:22268 entering state [proceeding][180] > 2012-06-27 20:55:14.939460 [NOTICE] sofia.c:5930 Ring-Ready sofia/internal/sip:1001@31.4.238.246:22268! > 2012-06-27 20:55:15.029459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001@31.4.238.246:22268 [BREAK] > 2012-06-27 20:55:15.029459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001@31.4.238.246:22268 [BREAK] > 2012-06-27 20:55:15.029459 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001@31.4.238.246:22268 entering state [proceeding][180] > 2012-06-27 20:55:15.069457 [DEBUG] switch_rtp.c:3527 Correct ip/port confirmed. > > When user 1001 answer the call, JitterBuffer is paused (in bold red): > > 2012-06-27 20:55:26.079458 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001@10.10.57.82:50203 [BREAK] > 2012-06-27 20:55:26.079458 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001@10.10.57.82:50203 [BREAK] > 2012-06-27 20:55:26.079458 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001@10.10.57.82:50203 entering state [completing][200] > 2012-06-27 20:55:26.079458 [DEBUG] sofia.c:5849 Remote SDP: > v=0 > o=- 3549812097 3549812099 IN IP4 10.10.57.82 > s=cpc_med > c=IN IP4 10.10.57.82 > b=AS:226 > t=0 0 > m=audio 22010 RTP/AVP 99 101 > a=rtpmap:99 iLBC/8000 > a=fmtp:99 mode=30 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > m=video 0 RTP/AVP 103 > c=IN IP4 10.10.57.82 > b=AS:200000 > a=rtpmap:103 H264/90000 > > 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3941 Looking for zrtp-hash > 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3919 Deciding whether to pass zrtp-hash between legs > 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3921 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash > 2012-06-27 20:55:26.079458 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001@10.10.57.82:50203 [BREAK] > 2012-06-27 20:55:26.079458 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/sip:1001@10.10.57.82:50203 [BREAK] > 2012-06-27 20:55:26.079458 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:1001@10.10.57.82:50203 entering state [ready][200] > 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3941 Looking for zrtp-hash > 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3919 Deciding whether to pass zrtp-hash between legs > 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3921 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash > 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5034 Audio Codec Compare [iLBC:99:8000:30:0]/[SILK:119:16000:20:30000] > 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5034 Audio Codec Compare [telephone-event:101:8000:20:0]/[SILK:119:16000:20:30000] > 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5148 Set 2833 dtmf send payload to 101 > 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5034 Audio Codec Compare [iLBC:99:8000:30:0]/[iLBC:97:8000:30:13330] > 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3020 Set Codec sofia/internal/sip:1001@10.10.57.82:50203 iLBC/8000 30 ms 240 samples 13330 bits > 2012-06-27 20:55:26.079458 [DEBUG] switch_core_codec.c:111 sofia/internal/sip:1001@10.10.57.82:50203 Original read codec set to iLBC:97 > 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:5148 Set 2833 dtmf send payload to 101 > 2012-06-27 20:55:26.079458 [DEBUG] sofia_glue.c:3269 AUDIO RTP [sofia/internal/sip:1001@10.10.57.82:50203] 79.150.184.57 port 10388 -> 10.10.57.82 port 22010 codec: 99 ms: 30 > 2012-06-27 20:55:26.079458 [DEBUG] switch_rtp.c:1861 Starting timer [soft] 240 bytes per 30ms > 2012-06-27 20:55:26.089461 [DEBUG] switch_rtp.c:2207 RTCP send rate is: 5000 and packet rate is: 30000 Remote Port: 22011 > 2012-06-27 20:55:26.089461 [DEBUG] switch_rtp.c:1230 Setting RTCP remote addr to 10.10.57.82:22011 > 2012-06-27 20:55:26.089461 [DEBUG] sofia_glue.c:3491 Setting Jitterbuffer to 90ms (3 frames) > 2012-06-27 20:55:26.089461 [DEBUG] sofia_glue.c:3533 Set 2833 dtmf send payload to 101 > 2012-06-27 20:55:26.089461 [DEBUG] sofia_glue.c:3539 Set 2833 dtmf receive payload to 101 > 2012-06-27 20:55:26.089461 [DEBUG] sofia_glue.c:3566 sofia/internal/sip:1001@10.10.57.82:50203 Set rtp dtmf delay to 40 > 2012-06-27 20:55:26.089461 [DEBUG] switch_channel.c:3301 (sofia/internal/sip:1001@10.10.57.82:50203) Callstate Change RINGING -> ACTIVE > 2012-06-27 20:55:26.089461 [DEBUG] switch_channel.c:3313 Send signal sofia/internal/1002@sukmad.dyndns-home.com:58180 [BREAK] > 2012-06-27 20:55:26.089461 [NOTICE] sofia.c:6573 Channel [sofia/internal/sip:1001@10.10.57.82:50203] has been answered > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_codec.c:244 sofia/internal/1002@sukmad.dyndns-home.com:58180 Restore previous codec iLBC:97. > 2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:2899 (sofia/internal/sip:1001@192.168.0.11:50199) Callstate Change RINGING -> HANGUP > 2012-06-27 20:55:26.109459 [NOTICE] switch_ivr_originate.c:3257 Hangup sofia/internal/sip:1001@192.168.0.11:50199 [CS_CONSUME_MEDIA] [LOSE_RACE] > 2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:2922 Send signal sofia/internal/sip:1001@192.168.0.11:50199 [KILL] > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001@192.168.0.11:50199 [BREAK] > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001@192.168.0.11:50199) Running State Change CS_HANGUP > 2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:2899 (sofia/internal/sip:1001@31.4.238.246:22268) Callstate Change RINGING -> HANGUP > 2012-06-27 20:55:26.109459 [NOTICE] switch_ivr_originate.c:3257 Hangup sofia/internal/sip:1001@31.4.238.246:22268 [CS_CONSUME_MEDIA] [LOSE_RACE] > 2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:2922 Send signal sofia/internal/sip:1001@31.4.238.246:22268 [KILL] > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:1001@192.168.0.11:50199) State HANGUP > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001@31.4.238.246:22268 [BREAK] > 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:469 Channel sofia/internal/sip:1001@192.168.0.11:50199 hanging up, cause: LOSE_RACE > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001@31.4.238.246:22268) Running State Change CS_HANGUP > 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:527 Sending CANCEL to sofia/internal/sip:1001@192.168.0.11:50199 > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:1001@192.168.0.11:50199 Standard HANGUP, cause: LOSE_RACE > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:1001@192.168.0.11:50199) State HANGUP going to sleep > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/sip:1001@192.168.0.11:50199) State Change CS_HANGUP -> CS_REPORTING > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001@192.168.0.11:50199 [BREAK] > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001@192.168.0.11:50199) Running State Change CS_REPORTING > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:1001@192.168.0.11:50199) State REPORTING > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:1001@192.168.0.11:50199 Standard REPORTING, cause: LOSE_RACE > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:1001@192.168.0.11:50199) State REPORTING going to sleep > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:1001@31.4.238.246:22268) State HANGUP > 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:469 Channel sofia/internal/sip:1001@31.4.238.246:22268 hanging up, cause: LOSE_RACE > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1001@192.168.0.11:50199) State Change CS_REPORTING -> CS_DESTROY > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001@192.168.0.11:50199 [BREAK] > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1429 Session 15 (sofia/internal/sip:1001@192.168.0.11:50199) Locked, Waiting on external entities > 2012-06-27 20:55:26.109459 [INFO] switch_nat.c:590 NAT port mapping disabled > 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:527 Sending CANCEL to sofia/internal/sip:1001@31.4.238.246:22268 > 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:754 Local SDP sofia/internal/1002@sukmad.dyndns-home.com:58180: > v=0 > o=FreeSWITCH 1340812160 1340812162 IN IP4 79.150.184.57 > s=FreeSWITCH > c=IN IP4 79.150.184.57 > t=0 0 > m=audio 11154 RTP/AVP 98 101 > a=rtpmap:98 iLBC/8000 > a=fmtp:98 mode=30 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:30 > a=sendrecv > m=video 10766 RTP/AVP 34 103 > a=rtpmap:34 H263/90000 > a=rtpmap:103 H264/90000 > > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:1001@31.4.238.246:22268 Standard HANGUP, cause: LOSE_RACE > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/sip:1001@31.4.238.246:22268) State HANGUP going to sleep > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/sip:1001@31.4.238.246:22268) State Change CS_HANGUP -> CS_REPORTING > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001@31.4.238.246:22268 [BREAK] > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001@31.4.238.246:22268) Running State Change CS_REPORTING > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/1002@sukmad.dyndns-home.com:58180 [BREAK] > 2012-06-27 20:55:26.109459 [DEBUG] switch_channel.c:3301 (sofia/internal/1002@sukmad.dyndns-home.com:58180) Callstate Change EARLY -> ACTIVE > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:1001@31.4.238.246:22268) State REPORTING > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:1001@31.4.238.246:22268 Standard REPORTING, cause: LOSE_RACE > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:685 (sofia/internal/sip:1001@31.4.238.246:22268) State REPORTING going to sleep > 2012-06-27 20:55:26.109459 [NOTICE] switch_ivr_originate.c:3284 Channel [sofia/internal/1002@sukmad.dyndns-home.com:58180] has been answered > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1001@31.4.238.246:22268) State Change CS_REPORTING -> CS_DESTROY > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001@31.4.238.246:22268 [BREAK] > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1429 Session 17 (sofia/internal/sip:1001@31.4.238.246:22268) Locked, Waiting on external entities > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002@sukmad.dyndns-home.com:58180 [BREAK] > 2012-06-27 20:55:26.109459 [DEBUG] sofia.c:5838 Channel sofia/internal/1002@sukmad.dyndns-home.com:58180 entering state [completed][200] > 2012-06-27 20:55:26.109459 [DEBUG] switch_ivr_originate.c:3341 Originate Resulted in Success: [sofia/internal/sip:1001@10.10.57.82:50203] > 2012-06-27 20:55:26.109459 [NOTICE] switch_core_session.c:1447 Session 15 (sofia/internal/sip:1001@192.168.0.11:50199) Ended > 2012-06-27 20:55:26.109459 [NOTICE] switch_core_session.c:1447 Session 17 (sofia/internal/sip:1001@31.4.238.246:22268) Ended > 2012-06-27 20:55:26.109459 [NOTICE] switch_core_session.c:1449 Close Channel sofia/internal/sip:1001@192.168.0.11:50199 [CS_DESTROY] > 2012-06-27 20:55:26.109459 [NOTICE] switch_core_session.c:1449 Close Channel sofia/internal/sip:1001@31.4.238.246:22268 [CS_DESTROY] > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:514 (sofia/internal/sip:1001@31.4.238.246:22268) Callstate Change HANGUP -> DOWN > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:514 (sofia/internal/sip:1001@192.168.0.11:50199) Callstate Change HANGUP -> DOWN > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:517 (sofia/internal/sip:1001@31.4.238.246:22268) Running State Change CS_DESTROY > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:1001@31.4.238.246:22268) State DESTROY > 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:374 sofia/internal/sip:1001@31.4.238.246:22268 SOFIA DESTROY > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:1001@31.4.238.246:22268 Standard DESTROY > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:517 (sofia/internal/sip:1001@192.168.0.11:50199) Running State Change CS_DESTROY > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:1001@31.4.238.246:22268) State DESTROY going to sleep > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:1001@192.168.0.11:50199) State DESTROY > 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:374 sofia/internal/sip:1001@192.168.0.11:50199 SOFIA DESTROY > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:1001@192.168.0.11:50199 Standard DESTROY > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:527 (sofia/internal/sip:1001@192.168.0.11:50199) State DESTROY going to sleep > 2012-06-27 20:55:26.109459 [DEBUG] switch_ivr_originate.c:3341 Originate Resulted in Success: [sofia/internal/sip:1001@10.10.57.82:50203] > 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:1603 sofia/internal/sip:1001@10.10.57.82:50203 PAUSE Jitterbuffer > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/sip:1001@10.10.57.82:50203 [BREAK] > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/1002@sukmad.dyndns-home.com:58180 [BREAK] > 2012-06-27 20:55:26.109459 [DEBUG] switch_ivr_bridge.c:1359 (sofia/internal/sip:1001@10.10.57.82:50203) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:1229 Send signal sofia/internal/sip:1001@10.10.57.82:50203 [BREAK] > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:1001@10.10.57.82:50203) Running State Change CS_EXCHANGE_MEDIA > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_state_machine.c:443 (sofia/internal/sip:1001@10.10.57.82:50203) State EXCHANGE_MEDIA > 2012-06-27 20:55:26.109459 [DEBUG] mod_sofia.c:582 SOFIA EXCHANGE_MEDIA > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002@sukmad.dyndns-home.com:58180 [BREAK] > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002@sukmad.dyndns-home.com:58180 [BREAK] > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:924 Send signal sofia/internal/1002@sukmad.dyndns-home.com:58180 [BREAK] > 2012-06-27 20:55:26.109459 [DEBUG] sofia.c:5838 Channel sofia/internal/1002@sukmad.dyndns-home.com:58180 entering state [ready][200] > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:840 Send signal sofia/internal/sip:1001@10.10.57.82:50203 [BREAK] > 2012-06-27 20:55:26.109459 [DEBUG] switch_core_session.c:840 Send signal sofia/internal/1002@sukmad.dyndns-home.com:58180 [BREAK] > 2012-06-27 20:55:26.469460 [INFO] switch_rtp.c:3510 Auto Changing port from 10.10.57.82:22010 to 31.4.238.246:22267 > 2012-06-27 20:55:26.469460 [DEBUG] switch_rtp.c:1230 Setting RTCP remote addr to 31.4.238.246:22268 > 2012-06-27 20:55:38.009460 [DEBUG] sofia_reg.c:371 Registered sip.netelip.com<http://sip.netelip.com> > > > I try a lot of test and all of them have the same issue, Why JitterBuffer is paused?. Is it possible to force JitterBuffer? > > > Thanks in advance, > > Regards > > > > !DSPAM:4feb659c32769194324459! > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org<mailto: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(a)lists.freeswitch.org<mailto: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 > > > !DSPAM:4feb659c32769194324459! > > >

1 0

Re: [Freeswitch-users] multiple registerations coudln't worked correctly!
by Peter Olsson 28 Jun '12

28 Jun '12

I still don't understand.. :) How could UserA use UserB's account? Aren't you using different passwords for the user? /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Samira Mh Skickat: den 28 juni 2012 10:05 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] multiple registerations coudln't worked correctly! i have implemented VODSL service using freeswitch,and create accounts for user on the server,but i don't want the userA use the account of userB, i want to restrict each user only use her/his account not able to use each other account that don't belong to his/her... also i don't want to restrict this using ip addresses ,want to limit using account ________________________________ From: Peter Olsson <peter.olsson(a)visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Sent: Thursday, June 28, 2012 9:44 AM Subject: Re: [Freeswitch-users] multiple registerations coudln't worked correctly! The option is to enable multiple-registrations, which will make both registrations work simultaneously. Why is this even a problem, what problem are you trying to solve? /Peter 28 jun 2012 kl. 06:21 skrev "Samira Mh" <saami_mh(a)ymail.com<mailto:saami_mh@ymail.com><mailto:saami_mh@ymail.com<mailto:saami_mh@ymail.com>>>: is there any solutions for the second registration couldn't overwrite the first one registeration ? ________________________________ From: Peter Olsson <peter.olsson(a)visionutveckling.se<mailto:peter.olsson@visionutveckling.se><mailto:peter.olsson@visionutveckling.se<mailto:peter.olsson@visionutveckling.se>>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org><mailto:freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>>> Sent: Wednesday, June 27, 2012 11:46 PM Subject: Re: [Freeswitch-users] multiple registerations coudln't worked correctly! If they end up with only one registration within FS it's just normal. Since the second registration will overwrite the first. /Peter ________________________________ Från: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org><mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>> [freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org><mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>>] för Samira Mh [saami_mh(a)ymail.com<mailto:saami_mh@ymail.com><mailto:saami_mh@ymail.com<mailto:saami_mh@ymail.com>>] Skickat: den 27 juni 2012 21:05 Till: Free SWITCH Users Help Ämne: [Freeswitch-users] multiple registerations coudln't worked correctly! hi, i have set <param name="multiple-registrations" value="false"/> on internal.xml and switch.conf.xml then restart freeswitch but I could register 2 sip phones using the same extension number at the same time! but i want to only one extension could be register using the same extensions plz help _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org><mailto:consulting@freeswitch.org<mailto: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(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org><mailto:FreeSWITCH-users@lists.freeswitch.org<mailto: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<http://www.freeswitch.org/><http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org><mailto:consulting@freeswitch.org<mailto:consulting@freeswitch.org>> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com<http://www.cudatel.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://wiki.freeswitch.org<http://wiki.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org><mailto:FreeSWITCH-users@lists.freeswitch.org<mailto: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<http://www.freeswitch.org/> !DSPAM:4febd86932764854317000! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com<http://www.cudatel.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://wiki.freeswitch.org<http://wiki.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto: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<http://www.freeswitch.org/> !DSPAM:4fec0eac32768939220457!

3 4

Building Debian packages for FS and 3rd-party libs
by Julian Pawlowski 28 Jun '12

28 Jun '12

Hi folks, I'm currently working on the system integration part of the GemeinschaftPBX version 5.0 (see github.com/jpawlowski/GBE) As compilation of FreeSWITCH and all dependent 3rd party libraries takes major time I would like to start using Debian packages which I can prepare once and reuse during the build phase of the appliance ISO. I understand your own Debian repository is not ready yet (or at least it's not public) so I wanted to implement a job on our build system (running Jenkins CI) to have a continuous build of FS packages based on changes in the FS Git. I started by reading documents in /freeswitch/debian/, especially README.source but as I am new to create Debian packages I can't work around this error when I try those command mentioned in the readme: --------- freeswitch_1.2~rc3~n20120628T140917Z.orig.tar.xz does not exist, creating from 'master' Building upstream tarball with compression 'xz -1v' 100 % 29.1 MiB / 143.4 MiB = 0.203 15 MiB/s 0:09 git-pbuilder ['-b', '-us', '-uc'] [] Building for distribution squeeze Base directory /var/cache/pbuilder/base-squeeze.cow does not exist git-pbuilder returned 1 Couldn't run 'git-pbuilder -b -us -uc' --------- I wasn't able to find anything useful regarding missing base-squeeze.cow file for pbuilder and would appreciate any assistance here. In addition, as I plan to have a full set of available modules for FS in our appliance distribution: Are there already Debian packages available for the needed 3rd-party modules like all the stuff for mod_freetdm (especially Sangoma parts)? Just wanted to ask before I reinvent the wheel ;-) Cheers, Julian

1 1

about FS symbols
by simon jin 28 Jun '12

28 Jun '12

HI everyone i have compiled the FS with CFLAGS="-g -ggdb -finstrument-functions" and MOD_CFLAGS="-g -ggdb -finstrument-functions", and it works done. but the problem comes when i use addr2line command to translate some function address to its name. This is the detail: $ addr2line -e freeswitch -f -a 0x2a955d3f8a 0x0000002a955d3f8a ?? ??:0 Note: 0x2a955d3f8a is some function's address I think that addr2line can NOT find freeswitch's symbols, but where it is, anyone any advice? thanks for your reply! yours simon

1 0

multiple registerations coudln't worked correctly!
by Samira Mh 28 Jun '12

28 Jun '12

hi, i have set <param name="multiple-registrations" value="false"/> on internal.xml and switch.conf.xml then restart freeswitch but I could register 2 sip phones using the same extension number at the same time! but i want to only one extension could be register using the same extensions plz help

2 4

Monitoring audio quality
by Michael Lutz 28 Jun '12

28 Jun '12

Hi Guys, I have a prety large system with a lot of traffic. Sometimes I get complains from customers about 'loud noises', 'not hearing audio', or just 'bad quality audio'. I can see a lot of info on the CDR's, no I have tried to search the wiki and groups for some explanaition about how to interpret them, but there is not much to find.. Is there someone who can tell me what these mean, and how to interpret them, could they help me in finding issues with audio? <rtp_audio_in_raw_bytes>573964</rtp_audio_in_raw_bytes> <rtp_audio_in_media_bytes>572932</rtp_audio_in_media_bytes> <rtp_audio_in_packet_count>3337</rtp_audio_in_packet_count> <rtp_audio_in_media_packet_count>3331</rtp_audio_in_media_packet_count> <rtp_audio_in_skip_packet_count>47</rtp_audio_in_skip_packet_count> <rtp_audio_in_jb_packet_count>0</rtp_audio_in_jb_packet_count> <rtp_audio_in_dtmf_packet_count>0</rtp_audio_in_dtmf_packet_count> <rtp_audio_in_cng_packet_count>0</rtp_audio_in_cng_packet_count> <rtp_audio_in_flush_packet_count>6</rtp_audio_in_flush_packet_count> <rtp_audio_in_largest_jb_size>0</rtp_audio_in_largest_jb_size> <rtp_audio_out_raw_bytes>485556</rtp_audio_out_raw_bytes> <rtp_audio_out_media_bytes>485556</rtp_audio_out_media_bytes> <rtp_audio_out_packet_count>2823</rtp_audio_out_packet_count> <rtp_audio_out_media_packet_count>2823</rtp_audio_out_media_packet_count> <rtp_audio_out_skip_packet_count>0</rtp_audio_out_skip_packet_count> <rtp_audio_out_dtmf_packet_count>0</rtp_audio_out_dtmf_packet_count> <rtp_audio_out_cng_packet_count>0</rtp_audio_out_cng_packet_count> <rtp_audio_rtcp_packet_count>0</rtp_audio_rtcp_packet_count> <rtp_audio_rtcp_octet_count>0</rtp_audio_rtcp_octet_count> Or are there other (better) methods of finding problems related to bad audio? Regards, Mike

9 15

how can run fs_cli commands in lua script?
by Samira Mh 27 Jun '12

27 Jun '12

hi all, i want to run the following command in lua script that is located on (/usr/local/freeswitch/scripts/lua1.lua) sofia_count_reg 1001-Q0ErXNX1RuYwPdctyuGeiA(a)public.gmane.org i ned to use the output of the above command how can i do that?

5 4

How to get last bridged call leg phone number?
by Gary Chen 27 Jun '12

27 Jun '12

FreeSWITCH Version 1.2.0-rc2+git~20120625T160048Z~690cae5de1 I am trying to setup hunt group using bridge app. I want it to ring the first available phone, if that person does not answer, then it should go to that person's voicemail. Here is my part of dialplan: <extension name="Local_Extension1"> <condition field="destination_number" expression="(^(5554447579$)"> <action application="set" data="ignore_early_media=true"/> <action application="bridge" data="{originate_continue_on_timeout=false}[ leg_timeout=10]sofia/internal/5554447580@192.168.1.22|[leg_timeout=10]sofia/internal/5554447578@192.168.1.22"/> </condition> <condition> Send call to voicemail ..... </condition> </extension> If no answer on 5554447580, I want it to goto 5554447580's voicemail. If no answer on 5554447578, I want it to goto 5554447578's voicemail. But I could not find the variable which contains the last bridged number information. How do I get the last dialed number in bridge app? I tried to add variable in each leg like following, but it did not work: <action application="bridge" data="{originate_continue_on_timeout=false}[ mailbox=7580,leg_timeout=10]sofia/internal/5554447580@192.168.1.22|[mailbox=7578,leg_timeout=10]sofia/internal/5554447578@192.168.1.22"/> Gary

2 6